Use BitrateAllocatorInterface in AudioSendStream and VideoSendStream
Followup to cl https://webrtc-review.googlesource.com/70880, which
introduced the interface.
Intended to enable tests using MockBitrateAllocator.
Bug: None
Change-Id: I0a784106acf37ff9aca118297233ebd2f2259ae4
Reviewed-on: https://webrtc-review.googlesource.com/c/107342
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25290}diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index c2b0d1a..f60f920 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -89,7 +89,7 @@
rtc::TaskQueue* worker_queue,
ProcessThread* module_process_thread,
RtpTransportControllerSendInterface* transport,
- BitrateAllocator* bitrate_allocator,
+ BitrateAllocatorInterface* bitrate_allocator,
RtcEventLog* event_log,
RtcpRttStats* rtcp_rtt_stats,
const absl::optional<RtpState>& suspended_rtp_state,
@@ -115,7 +115,7 @@
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
rtc::TaskQueue* worker_queue,
RtpTransportControllerSendInterface* transport,
- BitrateAllocator* bitrate_allocator,
+ BitrateAllocatorInterface* bitrate_allocator,
RtcEventLog* event_log,
RtcpRttStats* rtcp_rtt_stats,
const absl::optional<RtpState>& suspended_rtp_state,
diff --git a/audio/audio_send_stream.h b/audio/audio_send_stream.h
index 1ea676b..46c68b3 100644
--- a/audio/audio_send_stream.h
+++ b/audio/audio_send_stream.h
@@ -46,7 +46,7 @@
rtc::TaskQueue* worker_queue,
ProcessThread* module_process_thread,
RtpTransportControllerSendInterface* transport,
- BitrateAllocator* bitrate_allocator,
+ BitrateAllocatorInterface* bitrate_allocator,
RtcEventLog* event_log,
RtcpRttStats* rtcp_rtt_stats,
const absl::optional<RtpState>& suspended_rtp_state,
@@ -56,7 +56,7 @@
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
rtc::TaskQueue* worker_queue,
RtpTransportControllerSendInterface* transport,
- BitrateAllocator* bitrate_allocator,
+ BitrateAllocatorInterface* bitrate_allocator,
RtcEventLog* event_log,
RtcpRttStats* rtcp_rtt_stats,
const absl::optional<RtpState>& suspended_rtp_state,
@@ -140,7 +140,7 @@
size_t encoder_num_channels_ = 0;
bool sending_ = false;
- BitrateAllocator* const bitrate_allocator_;
+ BitrateAllocatorInterface* const bitrate_allocator_;
RtpTransportControllerSendInterface* const transport_;
rtc::CriticalSection packet_loss_tracker_cs_;
diff --git a/video/video_send_stream.cc b/video/video_send_stream.cc
index 054d27e..7999312 100644
--- a/video/video_send_stream.cc
+++ b/video/video_send_stream.cc
@@ -62,7 +62,7 @@
rtc::TaskQueue* worker_queue,
CallStats* call_stats,
RtpTransportControllerSendInterface* transport,
- BitrateAllocator* bitrate_allocator,
+ BitrateAllocatorInterface* bitrate_allocator,
SendDelayStats* send_delay_stats,
RtcEventLog* event_log,
VideoSendStream::Config config,
diff --git a/video/video_send_stream.h b/video/video_send_stream.h
index 1ff64dc..22aa4b1 100644
--- a/video/video_send_stream.h
+++ b/video/video_send_stream.h
@@ -59,7 +59,7 @@
rtc::TaskQueue* worker_queue,
CallStats* call_stats,
RtpTransportControllerSendInterface* transport,
- BitrateAllocator* bitrate_allocator,
+ BitrateAllocatorInterface* bitrate_allocator,
SendDelayStats* send_delay_stats,
RtcEventLog* event_log,
VideoSendStream::Config config,
