Pass SdpAudioFormat through Channel, without converting to CodecInst
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2516993002
Cr-Commit-Position: refs/heads/master@{#16165}
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index e6fa12e..c23084e 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -1542,14 +1542,15 @@
RTC_DCHECK_GE(ch, 0);
RTC_DCHECK(call);
config_.rtp.remote_ssrc = remote_ssrc;
+ config_.rtp.local_ssrc = local_ssrc;
+ config_.rtp.transport_cc = use_transport_cc;
+ config_.rtp.nack.rtp_history_ms = use_nack ? kNackRtpHistoryMs : 0;
+ config_.rtp.extensions = extensions;
config_.rtcp_send_transport = rtcp_send_transport;
config_.voe_channel_id = ch;
config_.sync_group = sync_group;
config_.decoder_factory = decoder_factory;
- RecreateAudioReceiveStream(local_ssrc,
- use_transport_cc,
- use_nack,
- extensions);
+ RecreateAudioReceiveStream();
}
~WebRtcAudioReceiveStream() {
@@ -1559,27 +1560,40 @@
void RecreateAudioReceiveStream(uint32_t local_ssrc) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
- RecreateAudioReceiveStream(local_ssrc,
- config_.rtp.transport_cc,
- config_.rtp.nack.rtp_history_ms != 0,
- config_.rtp.extensions);
+ config_.rtp.local_ssrc = local_ssrc;
+ RecreateAudioReceiveStream();
}
void RecreateAudioReceiveStream(bool use_transport_cc, bool use_nack) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
- RecreateAudioReceiveStream(config_.rtp.local_ssrc,
- use_transport_cc,
- use_nack,
- config_.rtp.extensions);
+ config_.rtp.transport_cc = use_transport_cc;
+ config_.rtp.nack.rtp_history_ms = use_nack ? kNackRtpHistoryMs : 0;
+ RecreateAudioReceiveStream();
}
void RecreateAudioReceiveStream(
const std::vector<webrtc::RtpExtension>& extensions) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
- RecreateAudioReceiveStream(config_.rtp.local_ssrc,
- config_.rtp.transport_cc,
- config_.rtp.nack.rtp_history_ms != 0,
- extensions);
+ config_.rtp.extensions = extensions;
+ RecreateAudioReceiveStream();
+ }
+
+ // Set a new payload type -> decoder map. The new map must be a superset of
+ // the old one.
+ void RecreateAudioReceiveStream(
+ const std::map<int, webrtc::SdpAudioFormat>& decoder_map) {
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+ RTC_DCHECK([&] {
+ for (const auto& item : config_.decoder_map) {
+ auto it = decoder_map.find(item.first);
+ if (it == decoder_map.end() || *it != item) {
+ return false; // The old map isn't a subset of the new map.
+ }
+ }
+ return true;
+ }());
+ config_.decoder_map = decoder_map;
+ RecreateAudioReceiveStream();
}
webrtc::AudioReceiveStream::Stats GetStats() const {
@@ -1617,21 +1631,11 @@
}
private:
- void RecreateAudioReceiveStream(
- uint32_t local_ssrc,
- bool use_transport_cc,
- bool use_nack,
- const std::vector<webrtc::RtpExtension>& extensions) {
+ void RecreateAudioReceiveStream() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
if (stream_) {
call_->DestroyAudioReceiveStream(stream_);
- stream_ = nullptr;
}
- config_.rtp.local_ssrc = local_ssrc;
- config_.rtp.transport_cc = use_transport_cc;
- config_.rtp.nack.rtp_history_ms = use_nack ? kNackRtpHistoryMs : 0;
- config_.rtp.extensions = extensions;
- RTC_DCHECK(!stream_);
stream_ = call_->CreateAudioReceiveStream(config_);
RTC_CHECK(stream_);
SetPlayout(playout_);
@@ -1901,40 +1905,34 @@
return true;
}
+ // Create a payload type -> SdpAudioFormat map with all the decoders. Fail
+ // unless the factory claims to support all decoders.
+ std::map<int, webrtc::SdpAudioFormat> decoder_map;
+ for (const AudioCodec& codec : codecs) {
+ auto format = AudioCodecToSdpAudioFormat(codec);
+ if (!IsCodec(codec, "cn") && !IsCodec(codec, "telephone-event") &&
+ !engine()->decoder_factory_->IsSupportedDecoder(format)) {
+ LOG(LS_ERROR) << "Unsupported codec: " << format;
+ return false;
+ }
+ decoder_map.insert({codec.id, std::move(format)});
+ }
+
if (playout_) {
// Receive codecs can not be changed while playing. So we temporarily
// pause playout.
ChangePlayout(false);
}
- bool result = true;
- for (const AudioCodec& codec : new_codecs) {
- webrtc::CodecInst voe_codec = {0};
- if (WebRtcVoiceEngine::ToCodecInst(codec, &voe_codec)) {
- LOG(LS_INFO) << ToString(codec);
- voe_codec.pltype = codec.id;
- for (const auto& ch : recv_streams_) {
- if (engine()->voe()->codec()->SetRecPayloadType(
- ch.second->channel(), voe_codec) == -1) {
- LOG_RTCERR2(SetRecPayloadType, ch.second->channel(),
- ToString(voe_codec));
- result = false;
- }
- }
- } else {
- LOG(LS_WARNING) << "Unknown codec " << ToString(codec);
- result = false;
- break;
- }
+ for (auto& kv : recv_streams_) {
+ kv.second->RecreateAudioReceiveStream(decoder_map);
}
- if (result) {
- recv_codecs_ = codecs;
- }
+ recv_codecs_ = codecs;
if (desired_playout_ && !playout_) {
ChangePlayout(desired_playout_);
}
- return result;
+ return true;
}
// Utility function called from SetSendParameters() to extract current send
