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Rewrite WebRtcSession media tests as PeerConnection tests

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Bug: webrtc:8222
Change-Id: I782a3227e30de70eb8f6c26a48723cb3510a84ad
Reviewed-on: https://webrtc-review.googlesource.com/6640
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20364}
This commit is contained in:
Steve Anton
2017-10-19 16:11:30 -07:00
committed by Commit Bot
parent 930e1af76f
commit 3df5dcac9b
16 changed files with 1642 additions and 1310 deletions
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7
media/base/fakemediaengine.h
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@ -488,12 +488,11 @@ class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
if (it != local_sinks_.end()) {
RTC_CHECK(it->second->source() == source);
} else {
local_sinks_.insert(
std::make_pair(ssrc, new VoiceChannelAudioSink(source)));
local_sinks_.insert(std::make_pair(
ssrc, rtc::MakeUnique<VoiceChannelAudioSink>(source)));
}
} else {
if (it != local_sinks_.end()) {
delete it->second;
local_sinks_.erase(it);
}
}
@ -506,7 +505,7 @@ class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
std::map<uint32_t, double> output_scalings_;
std::vector<DtmfInfo> dtmf_info_queue_;
AudioOptions options_;
std::map<uint32_t, VoiceChannelAudioSink*> local_sinks_;
std::map<uint32_t, std::unique_ptr<VoiceChannelAudioSink>> local_sinks_;
std::unique_ptr<webrtc::AudioSinkInterface> sink_;
int max_bps_;
};