webrtc/call/call.cc - Issue 1414743004: Implement AudioSendStream::GetStats().

Keyboard Shortcuts

	File
u :	up to issue
j / k :	jump to file after / before current file
J / K :	jump to next file with a comment after / before current file
	Side-by-side diff
i :	toggle intra-line diffs
e :	expand all comments
c :	collapse all comments
s :	toggle showing all comments
n / p :	next / previous diff chunk or comment
N / P :	next / previous comment
<Up> / <Down> :	next / previous line

	Issue
u :	up to list of issues
j / k :	jump to patch after / before current patch
o / <Enter> :	open current patch in side-by-side view
i :	open current patch in unified diff view

	Issue List
j / k :	jump to issue after / before current issue
o / <Enter> :	open current issue

Unified Diff: webrtc/call/call.cc

Issue 1414743004: Implement AudioSendStream::GetStats(). (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: workaround for android build issue Created 5 years, 2 months ago

Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.

Jump to:

Index: webrtc/call/call.cc

diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc

index cdb4f5d1a612ab3f8e1f6019e0d1e97abf573efa..eda209a01e76d826e3f75783101d4e33478153a1 100644

--- a/webrtc/call/call.cc

+++ b/webrtc/call/call.cc

@@ -145,7 +145,6 @@ Call::Call(const Call::Config& config)

network_enabled_(true),

receive_crit_(RWLockWrapper::CreateRWLock()),

send_crit_(RWLockWrapper::CreateRWLock()) {

- RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());

RTC_DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0);

RTC_DCHECK_GE(config.bitrate_config.start_bitrate_bps,

config.bitrate_config.min_bitrate_bps);

@@ -199,7 +198,8 @@ webrtc::AudioSendStream* Call::CreateAudioSendStream(

const webrtc::AudioSendStream::Config& config) {

TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream");

RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());

- AudioSendStream* send_stream = new AudioSendStream(config);

+ AudioSendStream* send_stream =

+ new AudioSendStream(config, config_.voice_engine);

if (!network_enabled_)

send_stream->SignalNetworkState(kNetworkDown);

{

« no previous file with comments | « webrtc/audio_send_stream.h ('k') | webrtc/test/fake_voice_engine.h » ('j') | no next file with comments »