Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public.
BUG=None
Review-Url: https://codereview.webrtc.org/2987763003
Cr-Commit-Position: refs/heads/master@{#19149}
diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc
index d9829c7..be5c18a 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -294,7 +294,7 @@
if (send_stream) {
VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
std::unique_ptr<voe::ChannelProxy> send_channel_proxy =
- voe_impl->GetChannelProxy(send_stream->config().voe_channel_id);
+ voe_impl->GetChannelProxy(send_stream->GetConfig().voe_channel_id);
channel_proxy_->AssociateSendChannel(*send_channel_proxy.get());
} else {
channel_proxy_->DisassociateSendChannel();
diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc
index 103d630..376af81 100644
--- a/webrtc/audio/audio_send_stream.cc
+++ b/webrtc/audio/audio_send_stream.cc
@@ -126,6 +126,11 @@
channel_proxy_->SetRtcpRttStats(nullptr);
}
+const webrtc::AudioSendStream::Config& AudioSendStream::GetConfig() const {
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+ return config_;
+}
+
void AudioSendStream::Reconfigure(
const webrtc::AudioSendStream::Config& new_config) {
ConfigureStream(this, new_config, false);
@@ -405,11 +410,6 @@
}
}
-const webrtc::AudioSendStream::Config& AudioSendStream::config() const {
- RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
- return config_;
-}
-
void AudioSendStream::SetTransportOverhead(int transport_overhead_per_packet) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
transport_->send_side_cc()->SetTransportOverhead(
diff --git a/webrtc/audio/audio_send_stream.h b/webrtc/audio/audio_send_stream.h
index 42a04ae..f46608d 100644
--- a/webrtc/audio/audio_send_stream.h
+++ b/webrtc/audio/audio_send_stream.h
@@ -50,8 +50,8 @@
~AudioSendStream() override;
// webrtc::AudioSendStream implementation.
+ const webrtc::AudioSendStream::Config& GetConfig() const override;
void Reconfigure(const webrtc::AudioSendStream::Config& config) override;
-
void Start() override;
void Stop() override;
bool SendTelephoneEvent(int payload_type, int payload_frequency, int event,
@@ -73,7 +73,6 @@
void OnPacketFeedbackVector(
const std::vector<PacketFeedback>& packet_feedback_vector) override;
- const webrtc::AudioSendStream::Config& config() const;
void SetTransportOverhead(int transport_overhead_per_packet);
RtpState GetRtpState() const;
diff --git a/webrtc/call/audio_send_stream.h b/webrtc/call/audio_send_stream.h
index 26729e4..fa5b5ee 100644
--- a/webrtc/call/audio_send_stream.h
+++ b/webrtc/call/audio_send_stream.h
@@ -129,6 +129,10 @@
rtc::scoped_refptr<AudioEncoderFactory> encoder_factory;
};
+ virtual ~AudioSendStream() = default;
+
+ virtual const webrtc::AudioSendStream::Config& GetConfig() const = 0;
+
// Reconfigure the stream according to the Configuration.
virtual void Reconfigure(const Config& config) = 0;
@@ -146,9 +150,6 @@
virtual void SetMuted(bool muted) = 0;
virtual Stats GetStats() const = 0;
-
- protected:
- virtual ~AudioSendStream() {}
};
} // namespace webrtc
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 6475e00..26d97aa 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -637,9 +637,9 @@
send_stream->Stop();
+ const uint32_t ssrc = send_stream->GetConfig().rtp.ssrc;
webrtc::internal::AudioSendStream* audio_send_stream =
static_cast<webrtc::internal::AudioSendStream*>(send_stream);
- const uint32_t ssrc = audio_send_stream->config().rtp.ssrc;
suspended_audio_send_ssrcs_[ssrc] = audio_send_stream->GetRtpState();
{
WriteLockScoped write_lock(*send_crit_);
@@ -656,7 +656,7 @@
}
UpdateAggregateNetworkState();
sent_rtp_audio_timer_ms_.Extend(audio_send_stream->GetActiveLifetime());
- delete audio_send_stream;
+ delete send_stream;
}
webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(
diff --git a/webrtc/media/engine/fakewebrtccall.h b/webrtc/media/engine/fakewebrtccall.h
index 15f2108..e4bc656 100644
--- a/webrtc/media/engine/fakewebrtccall.h
+++ b/webrtc/media/engine/fakewebrtccall.h
@@ -47,7 +47,7 @@
int id, const webrtc::AudioSendStream::Config& config);
int id() const { return id_; }
- const webrtc::AudioSendStream::Config& GetConfig() const;
+ const webrtc::AudioSendStream::Config& GetConfig() const override;
void SetStats(const webrtc::AudioSendStream::Stats& stats);
TelephoneEvent GetLatestTelephoneEvent() const;
bool IsSending() const { return sending_; }
