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(Auto)update libjingle 77263371-> 77296420

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git-svn-id: http://webrtc.googlecode.com/svn/trunk@7400 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
buildbot@webrtc.org
2014-10-08 22:24:30 +00:00
parent 536eb98408
commit 97abeee282
14 changed files with 6 additions and 77 deletions
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9
talk/media/base/fakemediaengine.h
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@@ -281,7 +281,6 @@ class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
}
return set_sending(flag != SEND_NOTHING);
}
virtual bool SetStartSendBandwidth(int bps) { return true; }
virtual bool SetMaxSendBandwidth(int bps) { return true; }
virtual bool AddRecvStream(const StreamParams& sp) {
if (!RtpHelper<VoiceMediaChannel>::AddRecvStream(sp))
@@ -477,7 +476,6 @@ class FakeVideoMediaChannel : public RtpHelper<VideoMediaChannel> {
: engine_(engine),
sent_intra_frame_(false),
requested_intra_frame_(false),
start_bps_(-1),
max_bps_(-1) {}
~FakeVideoMediaChannel();
@@ -489,7 +487,6 @@ class FakeVideoMediaChannel : public RtpHelper<VideoMediaChannel> {
const std::map<uint32, VideoRenderer*>& renderers() const {
return renderers_;
}
int start_bps() const { return start_bps_; }
int max_bps() const { return max_bps_; }
bool GetSendStreamFormat(uint32 ssrc, VideoFormat* format) {
if (send_formats_.find(ssrc) == send_formats_.end()) {
@@ -568,10 +565,6 @@ class FakeVideoMediaChannel : public RtpHelper<VideoMediaChannel> {
bool HasCapturer(uint32 ssrc) const {
return capturers_.find(ssrc) != capturers_.end();
}
virtual bool SetStartSendBandwidth(int bps) {
start_bps_ = bps;
return true;
}
virtual bool SetMaxSendBandwidth(int bps) {
max_bps_ = bps;
return true;
@@ -633,7 +626,6 @@ class FakeVideoMediaChannel : public RtpHelper<VideoMediaChannel> {
bool sent_intra_frame_;
bool requested_intra_frame_;
VideoOptions options_;
int start_bps_;
int max_bps_;
};
@@ -673,7 +665,6 @@ class FakeDataMediaChannel : public RtpHelper<DataMediaChannel> {
set_playout(receive);
return true;
}
virtual bool SetStartSendBandwidth(int bps) { return true; }
virtual bool SetMaxSendBandwidth(int bps) {
max_bps_ = bps;
return true;

2
talk/media/base/filemediaengine.h
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@@ -241,7 +241,6 @@ class FileVoiceChannel : public VoiceMediaChannel {
virtual bool AddRecvStream(const StreamParams& sp) { return true; }
virtual bool RemoveRecvStream(uint32 ssrc) { return true; }
virtual bool MuteStream(uint32 ssrc, bool on) { return false; }
virtual bool SetStartSendBandwidth(int bps) { return true; }
virtual bool SetMaxSendBandwidth(int bps) { return true; }
virtual bool SetOptions(const AudioOptions& options) {
options_ = options;
@@ -312,7 +311,6 @@ class FileVideoChannel : public VideoMediaChannel {
virtual bool AddRecvStream(const StreamParams& sp) { return true; }
virtual bool RemoveRecvStream(uint32 ssrc) { return true; }
virtual bool MuteStream(uint32 ssrc, bool on) { return false; }
virtual bool SetStartSendBandwidth(int bps) { return true; }
virtual bool SetMaxSendBandwidth(int bps) { return true; }
virtual bool SetOptions(const VideoOptions& options) {
options_ = options;

2
talk/media/base/mediachannel.h
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@@ -606,8 +606,6 @@ class MediaChannel : public sigslot::has_slots<> {
virtual int GetRtpSendTimeExtnId() const {
return -1;
}
// Sets the initial bandwidth to use when sending starts.
virtual bool SetStartSendBandwidth(int bps) = 0;
// Sets the maximum allowed bandwidth to use when sending data.
virtual bool SetMaxSendBandwidth(int bps) = 0;

1
talk/media/base/rtpdataengine.h
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@@ -96,7 +96,6 @@ class RtpDataMediaChannel : public DataMediaChannel {
timing_ = timing;
}
virtual bool SetStartSendBandwidth(int bps) { return true; }
virtual bool SetMaxSendBandwidth(int bps);
virtual bool SetRecvRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions) { return true; }

1
talk/media/base/videoengine_unittest.h
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@@ -1010,7 +1010,6 @@ class VideoMediaChannelTest : public testing::Test,
// Test that we can set the bandwidth.
void SetSendBandwidth() {
EXPECT_TRUE(channel_->SetStartSendBandwidth(64 * 1024));
EXPECT_TRUE(channel_->SetMaxSendBandwidth(-1)); // <= 0 means unlimited.
EXPECT_TRUE(channel_->SetMaxSendBandwidth(128 * 1024));
}

1
talk/media/other/linphonemediaengine.h
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@@ -143,7 +143,6 @@ class LinphoneVoiceChannel : public VoiceMediaChannel {
virtual void SetSendSsrc(uint32 id) {} // TODO: change RTP packet?
virtual bool SetRtcpCName(const std::string& cname) { return true; }
virtual bool Mute(bool on) { return mute_; }
virtual bool SetStartSendBandwidth(int bps) { return true; }
virtual bool SetMaxSendBandwidth(int bps) { return true; }
virtual bool SetOptions(int options) { return true; }
virtual bool SetRecvRtpHeaderExtensions(

1
talk/media/sctp/sctpdataengine.h
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@@ -172,7 +172,6 @@ class SctpDataMediaChannel : public DataMediaChannel,
// TODO(pthatcher): Cleanup MediaChannel interface, or at least
// don't try calling these and return false. Right now, things
// don't work if we return false.
virtual bool SetStartSendBandwidth(int bps) { return true; }
virtual bool SetMaxSendBandwidth(int bps) { return true; }
virtual bool SetRecvRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions) { return true; }

19
talk/media/webrtc/webrtcvideoengine.cc
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@@ -2904,25 +2904,6 @@ int WebRtcVideoMediaChannel::GetRtpSendTimeExtnId() const {
return -1;
}
bool WebRtcVideoMediaChannel::SetStartSendBandwidth(int bps) {
LOG(LS_INFO) << "WebRtcVideoMediaChannel::SetStartSendBandwidth";
if (!send_codec_) {
LOG(LS_INFO) << "The send codec has not been set up yet";
return true;
}
// On success, SetSendCodec() will reset |send_start_bitrate_| to |bps/1000|,
// by calling SanitizeBitrates. That method will also clamp the
// start bitrate between min and max, consistent with the override behavior
// in SetMaxSendBandwidth.
webrtc::VideoCodec new_codec = *send_codec_;
if (BitrateIsSet(bps)) {
new_codec.startBitrate = bps / 1000;
}
return SetSendCodec(new_codec);
}
bool WebRtcVideoMediaChannel::SetMaxSendBandwidth(int bps) {
LOG(LS_INFO) << "WebRtcVideoMediaChannel::SetMaxSendBandwidth";

1
talk/media/webrtc/webrtcvideoengine.h
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@@ -278,7 +278,6 @@ class WebRtcVideoMediaChannel : public rtc::MessageHandler,
virtual bool SetSendRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions);
virtual int GetRtpSendTimeExtnId() const;
virtual bool SetStartSendBandwidth(int bps);
virtual bool SetMaxSendBandwidth(int bps);
virtual bool SetOptions(const VideoOptions &options);
virtual bool GetOptions(VideoOptions *options) const {

6
talk/media/webrtc/webrtcvideoengine2.cc
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@@ -1230,12 +1230,6 @@ bool WebRtcVideoChannel2::SetSendRtpHeaderExtensions(
return true;
}
bool WebRtcVideoChannel2::SetStartSendBandwidth(int bps) {
// TODO(pbos): Implement.
LOG(LS_VERBOSE) << "SetStartSendBandwidth: " << bps;
return true;
}
bool WebRtcVideoChannel2::SetMaxSendBandwidth(int bps) {
// TODO(pbos): Implement.
LOG(LS_VERBOSE) << "SetMaxSendBandwidth: " << bps;

1
talk/media/webrtc/webrtcvideoengine2.h
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@@ -259,7 +259,6 @@ class WebRtcVideoChannel2 : public rtc::MessageHandler,
const std::vector<RtpHeaderExtension>& extensions) OVERRIDE;
virtual bool SetSendRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions) OVERRIDE;
virtual bool SetStartSendBandwidth(int bps) OVERRIDE;
virtual bool SetMaxSendBandwidth(int bps) OVERRIDE;
virtual bool SetOptions(const VideoOptions& options) OVERRIDE;
virtual bool GetOptions(VideoOptions* options) const OVERRIDE {

32
talk/media/webrtc/webrtcvideoengine_unittest.cc
View File

@@ -1488,44 +1488,24 @@ TEST_F(WebRtcVideoEngineTestFake, SetMaxBandwidthBelowMin) {
max_bandwidth_kbps, max_bandwidth_kbps, max_bandwidth_kbps);
}
// Test that the start bandwidth can be controlled separately from the max
// bandwidth.
TEST_F(WebRtcVideoEngineTestFake, SetStartBandwidth) {
EXPECT_TRUE(SetupEngine());
int channel_num = vie_.GetLastChannel();
EXPECT_TRUE(channel_->SetSendCodecs(engine_.codecs()));
int start_bandwidth_kbps = kStartBandwidthKbps + 1;
EXPECT_TRUE(channel_->SetStartSendBandwidth(start_bandwidth_kbps * 1000));
VerifyVP8SendCodec(channel_num, kVP8Codec.width, kVP8Codec.height, 0,
kMaxBandwidthKbps, kMinBandwidthKbps, start_bandwidth_kbps);
// Check that SetMaxSendBandwidth doesn't overwrite the start bandwidth.
int max_bandwidth_kbps = kMaxBandwidthKbps + 1;
EXPECT_TRUE(channel_->SetMaxSendBandwidth(max_bandwidth_kbps * 1000));
VerifyVP8SendCodec(channel_num, kVP8Codec.width, kVP8Codec.height, 0,
max_bandwidth_kbps, kMinBandwidthKbps, start_bandwidth_kbps);
}
// Test that the start bandwidth can be controlled by experiment.
// Test that the start bandwidth can be controlled by VideoOptions.
TEST_F(WebRtcVideoEngineTestFake, SetStartBandwidthOption) {
EXPECT_TRUE(SetupEngine());
int channel_num = vie_.GetLastChannel();
EXPECT_TRUE(channel_->SetSendCodecs(engine_.codecs()));
int start_bandwidth_kbps = kStartBandwidthKbps;
EXPECT_TRUE(channel_->SetStartSendBandwidth(start_bandwidth_kbps * 1000));
VerifyVP8SendCodec(channel_num, kVP8Codec.width, kVP8Codec.height, 0,
kMaxBandwidthKbps, kMinBandwidthKbps, start_bandwidth_kbps);
kMaxBandwidthKbps, kMinBandwidthKbps, kStartBandwidthKbps);
// Set the start bitrate option.
start_bandwidth_kbps = 1000;
int kBoostedStartBandwidthKbps = 1000;
ASSERT_NE(kStartBandwidthKbps, kBoostedStartBandwidthKbps);
cricket::VideoOptions options;
options.video_start_bitrate.Set(
start_bandwidth_kbps);
options.video_start_bitrate.Set(kBoostedStartBandwidthKbps);
EXPECT_TRUE(channel_->SetOptions(options));
// Check that start bitrate has changed to the new value.
VerifyVP8SendCodec(channel_num, kVP8Codec.width, kVP8Codec.height, 0,
kMaxBandwidthKbps, kMinBandwidthKbps, start_bandwidth_kbps);
kMaxBandwidthKbps, kMinBandwidthKbps, kBoostedStartBandwidthKbps);
}
// Test that SetMaxSendBandwidth works as expected in conference mode.

6
talk/media/webrtc/webrtcvoiceengine.cc
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@@ -3206,12 +3206,6 @@ bool WebRtcVoiceMediaChannel::MuteStream(uint32 ssrc, bool muted) {
return true;
}
bool WebRtcVoiceMediaChannel::SetStartSendBandwidth(int bps) {
// TODO(andresp): Add support for setting an independent start bandwidth when
// bandwidth estimation is enabled for voice engine.
return false;
}
bool WebRtcVoiceMediaChannel::SetMaxSendBandwidth(int bps) {
LOG(LS_INFO) << "WebRtcVoiceMediaChanne::SetSendBandwidth.";

1
talk/media/webrtc/webrtcvoiceengine.h
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@@ -374,7 +374,6 @@ class WebRtcVoiceMediaChannel
const rtc::PacketTime& packet_time);
virtual void OnReadyToSend(bool ready) {}
virtual bool MuteStream(uint32 ssrc, bool on);
virtual bool SetStartSendBandwidth(int bps);
virtual bool SetMaxSendBandwidth(int bps);
virtual bool GetStats(VoiceMediaInfo* info);
// Gets last reported error from WebRtc voice engine. This should be only