Add callback for send codec in audio too
It turns out there's a similar linkage as the one for video.
Tests are coming in https://webrtc-review.googlesource.com/c/src/+/307461
Bug: webrtc:13931
Change-Id: I638d1a1907116a71481aa88dce932492323ae5b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307463
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40206}
diff --git a/media/base/fake_media_engine.h b/media/base/fake_media_engine.h
index b00de20..74128cb 100644
--- a/media/base/fake_media_engine.h
+++ b/media/base/fake_media_engine.h
@@ -420,6 +420,9 @@
bool SenderNonSenderRttEnabled() const override { return false; }
void SetReceiveNackEnabled(bool enabled) {}
void SetReceiveNonSenderRttEnabled(bool enabled) {}
+ bool SendCodecHasNack() const override { return false; }
+ void SetSendCodecChangedCallback(
+ absl::AnyInvocable<void()> callback) override {}
private:
class VoiceChannelAudioSink : public AudioSource::Sink {
diff --git a/media/base/media_channel.h b/media/base/media_channel.h
index 9749cd0..e982284 100644
--- a/media/base/media_channel.h
+++ b/media/base/media_channel.h
@@ -247,9 +247,14 @@
webrtc::VideoEncoderFactory::EncoderSelectorInterface* encoder_selector) {
}
virtual webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const = 0;
+ virtual bool SendCodecHasNack() const = 0;
// Called whenever the list of sending SSRCs changes.
virtual void SetSsrcListChangedCallback(
absl::AnyInvocable<void(const std::set<uint32_t>&)> callback) = 0;
+ // TODO(bugs.webrtc.org/13931): Remove when configuration is more sensible
+ virtual void SetSendCodecChangedCallback(
+ absl::AnyInvocable<void()> callback) = 0;
+
// Get the underlying send/receive implementation channel for testing.
// TODO(bugs.webrtc.org/13931): Remove method and the fakes that depend on it.
virtual MediaChannel* ImplForTesting() = 0;
@@ -952,11 +957,7 @@
// Information queries to support SetReceiverFeedbackParameters
virtual webrtc::RtcpMode SendCodecRtcpMode() const = 0;
virtual bool SendCodecHasLntf() const = 0;
- virtual bool SendCodecHasNack() const = 0;
virtual absl::optional<int> SendCodecRtxTime() const = 0;
- // TODO(bugs.webrtc.org/13931): Remove when configuration is more sensible
- virtual void SetSendCodecChangedCallback(
- absl::AnyInvocable<void()> callback) = 0;
};
class VideoMediaReceiveChannelInterface : public MediaReceiveChannelInterface {
diff --git a/media/base/media_channel_impl.h b/media/base/media_channel_impl.h
index dfd9fb3..d4d8182 100644
--- a/media/base/media_channel_impl.h
+++ b/media/base/media_channel_impl.h
@@ -206,6 +206,8 @@
void OnNetworkRouteChanged(absl::string_view transport_name,
const rtc::NetworkRoute& network_route) override =
0;
+ void SetSendCodecChangedCallback(
+ absl::AnyInvocable<void()> callback) override = 0;
// Methods from the APIs that are implemented in MediaChannelUtil
using MediaChannelUtil::ExtmapAllowMixed;
@@ -469,6 +471,11 @@
bool SenderNonSenderRttEnabled() const override {
return impl_->SenderNonSenderRttEnabled();
}
+ bool SendCodecHasNack() const override { return impl()->SendCodecHasNack(); }
+ void SetSendCodecChangedCallback(
+ absl::AnyInvocable<void()> callback) override {
+ impl()->SetSendCodecChangedCallback(std::move(callback));
+ }
MediaChannel* ImplForTesting() override { return impl_; }
private:
diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index a763ddb..1b240d6 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -1761,15 +1761,17 @@
}
call_->GetTransportControllerSend()->SetSdpBitrateParameters(bitrate_config);
+ send_codecs_ = codecs;
// In legacy kBoth mode, the MediaChannel sets the NACK status.
// In other modes, this is done externally.
if (role() == MediaChannel::Role::kBoth) {
SetReceiveNackEnabled(send_codec_spec_->nack_enabled);
SetReceiveNonSenderRttEnabled(send_codec_spec_->enable_non_sender_rtt);
+ } else if (send_codec_changed_callback_) {
+ send_codec_changed_callback_();
}
- send_codecs_ = codecs;
return true;
}
diff --git a/media/engine/webrtc_voice_engine.h b/media/engine/webrtc_voice_engine.h
index e73c24c..474acc4 100644
--- a/media/engine/webrtc_voice_engine.h
+++ b/media/engine/webrtc_voice_engine.h
@@ -18,6 +18,7 @@
#include <memory>
#include <set>
#include <string>
+#include <utility>
#include <vector>
#include "absl/functional/any_invocable.h"
@@ -304,9 +305,14 @@
}
return send_codec_spec_->enable_non_sender_rtt;
}
+ bool SendCodecHasNack() const override { return SenderNackEnabled(); }
void SetReceiveNackEnabled(bool enabled) override;
void SetReceiveNonSenderRttEnabled(bool enabled) override;
+ void SetSendCodecChangedCallback(
+ absl::AnyInvocable<void()> callback) override {
+ send_codec_changed_callback_ = std::move(callback);
+ }
private:
bool SetOptions(const AudioOptions& options);
@@ -404,6 +410,9 @@
void FillSendCodecStats(VoiceMediaSendInfo* voice_media_info);
void FillReceiveCodecStats(VoiceMediaReceiveInfo* voice_media_info);
+ // Callback invoked whenever the send codec changes.
+ // TODO(bugs.webrtc.org/13931): Remove again when coupling isn't needed.
+ absl::AnyInvocable<void()> send_codec_changed_callback_;
// Callback invoked whenever the list of SSRCs changes.
absl::AnyInvocable<void(const std::set<uint32_t>&)>
ssrc_list_changed_callback_;
diff --git a/pc/channel.cc b/pc/channel.cc
index bf98278..f9ee3ff 100644
--- a/pc/channel.cc
+++ b/pc/channel.cc
@@ -886,7 +886,9 @@
crypto_options,
ssrc_generator),
send_channel_(media_channel_impl_->AsVoiceChannel()),
- receive_channel_(media_channel_impl_->AsVoiceChannel()) {}
+ receive_channel_(media_channel_impl_->AsVoiceChannel()) {
+ InitCallback();
+}
VoiceChannel::~VoiceChannel() {
TRACE_EVENT0("webrtc", "VoiceChannel::~VoiceChannel");
@@ -894,6 +896,18 @@
DisableMedia_w();
}
+void VoiceChannel::InitCallback() {
+ RTC_DCHECK_RUN_ON(worker_thread());
+ // TODO(bugs.webrtc.org/13931): Remove when values are set
+ // in a more sensible fashion
+ send_channel_.SetSendCodecChangedCallback([this]() {
+ RTC_DCHECK_RUN_ON(worker_thread());
+ // Adjust receive streams based on send codec.
+ receive_channel_.SetReceiveNackEnabled(send_channel_.SendCodecHasNack());
+ receive_channel_.SetReceiveNonSenderRttEnabled(
+ send_channel_.SenderNonSenderRttEnabled());
+ });
+}
void VoiceChannel::UpdateMediaSendRecvState_w() {
// Render incoming data if we're the active call, and we have the local
// content. We receive data on the default channel and multiplexed streams.
diff --git a/pc/channel.h b/pc/channel.h
index a84a842..89af890 100644
--- a/pc/channel.h
+++ b/pc/channel.h
@@ -435,6 +435,7 @@
}
private:
+ void InitCallback();
// overrides from BaseChannel
void UpdateMediaSendRecvState_w() RTC_RUN_ON(worker_thread()) override;
bool SetLocalContent_w(const MediaContentDescription* content,
diff --git a/pc/test/mock_voice_media_channel.h b/pc/test/mock_voice_media_channel.h
index 8748532..e3eb42d 100644
--- a/pc/test/mock_voice_media_channel.h
+++ b/pc/test/mock_voice_media_channel.h
@@ -67,10 +67,15 @@
ChooseReceiverReportSsrc,
(const std::set<uint32_t>&),
(override));
+ MOCK_METHOD(bool, SendCodecHasNack, (), (const, override));
MOCK_METHOD(void,
SetSsrcListChangedCallback,
(absl::AnyInvocable<void(const std::set<uint32_t>&)>),
(override));
+ MOCK_METHOD(void,
+ SetSendCodecChangedCallback,
+ (absl::AnyInvocable<void()>),
+ (override));
MOCK_METHOD(void, OnDemuxerCriteriaUpdatePending, (), (override));
MOCK_METHOD(void, OnDemuxerCriteriaUpdateComplete, (), (override));
MOCK_METHOD(int, GetRtpSendTimeExtnId, (), (const, override));
