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Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public.

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BUG=None

Review-Url: https://codereview.webrtc.org/2987763003
Cr-Commit-Position: refs/heads/master@{#19149}
This commit is contained in:
eladalon 2017-07-26 02:09:44 -07:00 committed by Commit Bot
parent 22d162dc61
commit abbc430ea0
6 changed files with 14 additions and 14 deletions
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2
webrtc/audio/audio_receive_stream.cc
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@ -294,7 +294,7 @@ void AudioReceiveStream::AssociateSendStream(AudioSendStream* send_stream) {
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();

10
webrtc/audio/audio_send_stream.cc
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@ -126,6 +126,11 @@ AudioSendStream::~AudioSendStream() {
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 @@ void AudioSendStream::OnPacketFeedbackVector(
}
}
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(

3
webrtc/audio/audio_send_stream.h
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@ -50,8 +50,8 @@ class AudioSendStream final : public webrtc::AudioSendStream,
~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 @@ class AudioSendStream final : public webrtc::AudioSendStream,
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;

7
webrtc/call/audio_send_stream.h
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@ -129,6 +129,10 @@ class AudioSendStream {
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 @@ class AudioSendStream {
virtual void SetMuted(bool muted) = 0;
virtual Stats GetStats() const = 0;
protected:
virtual ~AudioSendStream() {}
};
} // namespace webrtc

4
webrtc/call/call.cc
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@ -637,9 +637,9 @@ void Call::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
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 @@ void Call::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
}
UpdateAggregateNetworkState();
sent_rtp_audio_timer_ms_.Extend(audio_send_stream->GetActiveLifetime());
delete audio_send_stream;
delete send_stream;
}
webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(

2
webrtc/media/engine/fakewebrtccall.h
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@ -47,7 +47,7 @@ class FakeAudioSendStream final : public webrtc::AudioSendStream {
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_; }