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Adding getParameters/setParameters APIs to RtpReceiver.

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This is similar to how a "receive" method is used to apply
RtpParameters to an RtpReceiver in ORTC. Currently, SetParameters
doesn't allow changing the parameters, so the main use of the API is
to retrieve the set of configured codecs. But other uses will likely
be made possible in the future.

R=glaznev@webrtc.org, pthatcher@webrtc.org, tkchin@webrtc.org

Review URL: https://codereview.webrtc.org/1917193008 .

Cr-Commit-Position: refs/heads/master@{#12761}
This commit is contained in:
Taylor Brandstetter
2016-05-16 11:40:30 -07:00
parent 8bce67b745
commit db0cd9e774
33 changed files with 898 additions and 236 deletions
Download Patch File Download Diff File Expand all files Collapse all files

74
webrtc/media/engine/webrtcvoiceengine.cc
View File

@ -1423,13 +1423,13 @@ bool WebRtcVoiceMediaChannel::SetRecvParameters(
return true;
}
webrtc::RtpParameters WebRtcVoiceMediaChannel::GetRtpParameters(
webrtc::RtpParameters WebRtcVoiceMediaChannel::GetRtpSendParameters(
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.";
LOG(LS_WARNING) << "Attempting to get RTP send parameters for stream "
<< "with ssrc " << ssrc << " which doesn't exist.";
return webrtc::RtpParameters();
}
@ -1442,7 +1442,7 @@ webrtc::RtpParameters WebRtcVoiceMediaChannel::GetRtpParameters(
return rtp_params;
}
bool WebRtcVoiceMediaChannel::SetRtpParameters(
bool WebRtcVoiceMediaChannel::SetRtpSendParameters(
uint32_t ssrc,
const webrtc::RtpParameters& parameters) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
@ -1451,13 +1451,22 @@ bool WebRtcVoiceMediaChannel::SetRtpParameters(
}
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.";
LOG(LS_WARNING) << "Attempting to set RTP send 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.";
// TODO(deadbeef): Handle setting parameters with a list of codecs in a
// different order (which should change the send codec).
webrtc::RtpParameters current_parameters = GetRtpSendParameters(ssrc);
if (current_parameters.codecs != parameters.codecs) {
LOG(LS_ERROR) << "Using SetParameters to change the set of codecs "
<< "is not currently supported.";
return false;
}
if (!SetChannelSendParameters(it->second->channel(), parameters)) {
LOG(LS_WARNING) << "Failed to set send RtpParameters.";
return false;
}
// Codecs are handled at the WebRtcVoiceMediaChannel level.
@ -1467,6 +1476,47 @@ bool WebRtcVoiceMediaChannel::SetRtpParameters(
return true;
}
webrtc::RtpParameters WebRtcVoiceMediaChannel::GetRtpReceiveParameters(
uint32_t ssrc) const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
auto it = recv_streams_.find(ssrc);
if (it == recv_streams_.end()) {
LOG(LS_WARNING) << "Attempting to get RTP receive parameters for stream "
<< "with ssrc " << ssrc << " which doesn't exist.";
return webrtc::RtpParameters();
}
// TODO(deadbeef): Return stream-specific parameters.
webrtc::RtpParameters rtp_params = CreateRtpParametersWithOneEncoding();
for (const AudioCodec& codec : recv_codecs_) {
rtp_params.codecs.push_back(codec.ToCodecParameters());
}
return rtp_params;
}
bool WebRtcVoiceMediaChannel::SetRtpReceiveParameters(
uint32_t ssrc,
const webrtc::RtpParameters& parameters) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
if (!ValidateRtpParameters(parameters)) {
return false;
}
auto it = recv_streams_.find(ssrc);
if (it == recv_streams_.end()) {
LOG(LS_WARNING) << "Attempting to set RTP receive parameters for stream "
<< "with ssrc " << ssrc << " which doesn't exist.";
return false;
}
webrtc::RtpParameters current_parameters = GetRtpReceiveParameters(ssrc);
if (current_parameters != parameters) {
LOG(LS_ERROR) << "Changing the RTP receive parameters is currently "
<< "unsupported.";
return false;
}
return true;
}
bool WebRtcVoiceMediaChannel::ValidateRtpParameters(
const webrtc::RtpParameters& rtp_parameters) {
if (rtp_parameters.encodings.size() != 1) {
@ -1769,7 +1819,7 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
}
// TODO(solenberg): SetMaxSendBitrate() yields another call to SetSendCodec().
// Check if it is possible to fuse with the previous call in this function.
SetChannelParameters(channel, rtp_parameters);
SetChannelSendParameters(channel, rtp_parameters);
// Set the CN payloadtype and the VAD status.
if (send_codec_spec_.cng_payload_type != -1) {
@ -2369,15 +2419,15 @@ bool WebRtcVoiceMediaChannel::SetMaxSendBitrate(int bps) {
max_send_bitrate_bps_ = bps;
for (const auto& kv : send_streams_) {
if (!SetChannelParameters(kv.second->channel(),
kv.second->rtp_parameters())) {
if (!SetChannelSendParameters(kv.second->channel(),
kv.second->rtp_parameters())) {
return false;
}
}
return true;
}
bool WebRtcVoiceMediaChannel::SetChannelParameters(
bool WebRtcVoiceMediaChannel::SetChannelSendParameters(
int channel,
const webrtc::RtpParameters& parameters) {
RTC_CHECK_EQ(1UL, parameters.encodings.size());