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,
