| Index: webrtc/audio/audio_send_stream.h
|
| diff --git a/webrtc/audio/audio_send_stream.h b/webrtc/audio/audio_send_stream.h
|
| index 264f9b3a69f53d9c71e732919e10619e51edb26c..a993d5f2f9c03f4add6ea239a9123f3b7bae1a6c 100644
|
| --- a/webrtc/audio/audio_send_stream.h
|
| +++ b/webrtc/audio/audio_send_stream.h
|
| @@ -17,6 +17,7 @@
|
| #include "webrtc/audio_state.h"
|
| #include "webrtc/base/constructormagic.h"
|
| #include "webrtc/base/thread_checker.h"
|
| +#include "webrtc/call/bitrate_allocator.h"
|
|
|
| namespace webrtc {
|
| class CongestionController;
|
| @@ -27,11 +28,13 @@ class ChannelProxy;
|
| } // namespace voe
|
|
|
| namespace internal {
|
| -class AudioSendStream final : public webrtc::AudioSendStream {
|
| +class AudioSendStream final : public webrtc::AudioSendStream,
|
| + public webrtc::BitrateAllocatorObserver {
|
| public:
|
| AudioSendStream(const webrtc::AudioSendStream::Config& config,
|
| const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
|
| - CongestionController* congestion_controller);
|
| + CongestionController* congestion_controller,
|
| + BitrateAllocator* bitrate_allocator);
|
| ~AudioSendStream() override;
|
|
|
| // webrtc::AudioSendStream implementation.
|
| @@ -44,6 +47,12 @@ class AudioSendStream final : public webrtc::AudioSendStream {
|
|
|
| void SignalNetworkState(NetworkState state);
|
| bool DeliverRtcp(const uint8_t* packet, size_t length);
|
| +
|
| + // Implements BitrateAllocatorObserver.
|
| + uint32_t OnBitrateUpdated(uint32_t bitrate_bps,
|
| + uint8_t fraction_loss,
|
| + int64_t rtt) override;
|
| +
|
| const webrtc::AudioSendStream::Config& config() const;
|
|
|
| private:
|
| @@ -54,6 +63,8 @@ class AudioSendStream final : public webrtc::AudioSendStream {
|
| rtc::scoped_refptr<webrtc::AudioState> audio_state_;
|
| std::unique_ptr<voe::ChannelProxy> channel_proxy_;
|
|
|
| + BitrateAllocator* const bitrate_allocator_;
|
| +
|
| RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream);
|
| };
|
| } // namespace internal
|
|
|