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Fix issue with conflicting behavior if setting a max BW with b=AS on both audio and video.

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This reverts to previous behavior where b=AS only affects the codec bitrate for audio streams, and not the max bandwidth estimate.

BUG=chromium:703903

Review-Url: https://codereview.webrtc.org/2774123002
Cr-Commit-Position: refs/heads/master@{#17386}
This commit is contained in:
stefan
2017-03-27 03:51:18 -07:00
committed by Commit bot
parent 64f573bbb8
commit 1ccf73f830
3 changed files with 7 additions and 14 deletions
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14
webrtc/media/engine/webrtcvoiceengine.cc
View File

@ -1638,14 +1638,6 @@ bool WebRtcVoiceMediaChannel::SetSendParameters(
return false;
}
if (params.max_bandwidth_bps >= 0) {
// Note that max_bandwidth_bps intentionally takes priority over the
// bitrate config for the codec.
bitrate_config_.max_bitrate_bps =
params.max_bandwidth_bps == 0 ? -1 : params.max_bandwidth_bps;
}
call_->SetBitrateConfig(bitrate_config_);
if (!ValidateRtpExtensions(params.extensions)) {
return false;
}
@ -1917,6 +1909,7 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
// parameters.
// TODO(solenberg): Refactor this logic once we create AudioEncoders here.
webrtc::AudioSendStream::Config::SendCodecSpec send_codec_spec;
webrtc::Call::Config::BitrateConfig bitrate_config;
{
send_codec_spec.nack_enabled = send_codec_spec_.nack_enabled;
@ -1930,7 +1923,7 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
send_codec_spec.transport_cc_enabled = HasTransportCc(*codec);
send_codec_spec.nack_enabled = HasNack(*codec);
bitrate_config_ = GetBitrateConfigForCodec(*codec);
bitrate_config = GetBitrateConfigForCodec(*codec);
// For Opus as the send codec, we are to determine inband FEC, maximum
// playback rate, and opus internal dtx.
@ -2007,8 +2000,9 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
} else {
// If the codec isn't changing, set the start bitrate to -1 which means
// "unchanged" so that BWE isn't affected.
bitrate_config_.start_bitrate_bps = -1;
bitrate_config.start_bitrate_bps = -1;
}
call_->SetBitrateConfig(bitrate_config);
// Check if the transport cc feedback or NACK status has changed on the
// preferred send codec, and in that case reconfigure all receive streams.

1
webrtc/media/engine/webrtcvoiceengine.h
View File

@ -262,7 +262,6 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
bool playout_ = false;
bool send_ = false;
webrtc::Call* const call_ = nullptr;
webrtc::Call::Config::BitrateConfig bitrate_config_;
// Queue of unsignaled SSRCs; oldest at the beginning.
std::vector<uint32_t> unsignaled_recv_ssrcs_;

6
webrtc/media/engine/webrtcvoiceengine_unittest.cc
View File

@ -1533,16 +1533,16 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCapsMinAndStartBitrate) {
}
TEST_F(WebRtcVoiceEngineTestFake,
SetMaxSendBandwidthShouldPreserveOtherBitrates) {
SetMaxSendBandwidthForAudioDoesntAffectBwe) {
SetSendCodecsShouldWorkForBitrates("100", 100000, "150", 150000, "200",
200000);
send_parameters_.max_bandwidth_bps = 300000;
send_parameters_.max_bandwidth_bps = 100000;
SetSendParameters(send_parameters_);
EXPECT_EQ(100000, call_.GetConfig().bitrate_config.min_bitrate_bps)
<< "Setting max bitrate should keep previous min bitrate.";
EXPECT_EQ(-1, call_.GetConfig().bitrate_config.start_bitrate_bps)
<< "Setting max bitrate should not reset start bitrate.";
EXPECT_EQ(300000, call_.GetConfig().bitrate_config.max_bitrate_bps);
EXPECT_EQ(200000, call_.GetConfig().bitrate_config.max_bitrate_bps);
}
// Test that we can enable NACK with opus as caller.