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Use BitrateAllocatorInterface in AudioSendStream and VideoSendStream

Browse Source 
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}
This commit is contained in:
Niels Möller 2018-10-22 13:00:40 +02:00 committed by Commit Bot
parent ff292f30d9
commit 67b011d22c
4 changed files with 7 additions and 7 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

4
audio/audio_send_stream.cc
View File

@ -89,7 +89,7 @@ AudioSendStream::AudioSendStream(
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 @@ AudioSendStream::AudioSendStream(
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,

6
audio/audio_send_stream.h
View File

@ -46,7 +46,7 @@ class AudioSendStream final : public webrtc::AudioSendStream,
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 @@ class AudioSendStream final : public webrtc::AudioSendStream,
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 @@ class AudioSendStream final : public webrtc::AudioSendStream,
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_;

2
video/video_send_stream.cc
View File

@ -62,7 +62,7 @@ VideoSendStream::VideoSendStream(
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,

2
video/video_send_stream.h
View File

@ -59,7 +59,7 @@ class VideoSendStream : public webrtc::VideoSendStream {
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,