Chromium Code Reviews| Index: webrtc/voice_engine/channel.cc |
| diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc |
| index b0bae4e4ceb0e05cd1ca421efdbc5943cf640e19..35412ce41cec091b09f282e0ecf817cbe7d30d69 100644 |
| --- a/webrtc/voice_engine/channel.cc |
| +++ b/webrtc/voice_engine/channel.cc |
| @@ -1288,10 +1288,16 @@ bool Channel::SetEncoder(int payload_type, |
| // supposedly carries the sample rate, but only clock rate seems sensible to |
| // send to the RTP/RTCP module. |
| lies.plfreq = encoder->RtpTimestampRateHz(); |
| - lies.pacsize = 0; |
| + lies.pacsize = rtc::CheckedDivExact( |
| + static_cast<int>(encoder->Max10MsFramesInAPacket() * lies.plfreq), 100); |
| lies.channels = encoder->NumChannels(); |
| lies.rate = 0; |
| + // Store a similarily made-up CodecInst for GetSendCodec to return. |
|
henrika_webrtc
2017/06/09 13:50:04
Similar to what?
ossu
2017/06/09 13:54:00
On line 1281 I explain that I'm making a CodecInst
|
| + CodecInst send_codec = lies; |
| + send_codec.plfreq = encoder->SampleRateHz(); |
| + cached_send_codec_.emplace(send_codec); |
| + |
| if (_rtpRtcpModule->RegisterSendPayload(lies) != 0) { |
| _rtpRtcpModule->DeRegisterSendPayload(payload_type); |
| if (_rtpRtcpModule->RegisterSendPayload(lies) != 0) { |
| @@ -1343,18 +1349,16 @@ int32_t Channel::DeRegisterVoiceEngineObserver() { |
| } |
| int32_t Channel::GetSendCodec(CodecInst& codec) { |
| - { |
| + if (cached_send_codec_) { |
| + codec = *cached_send_codec_; |
| + return 0; |
| + } else { |
| const CodecInst* send_codec = codec_manager_.GetCodecInst(); |
| if (send_codec) { |
| codec = *send_codec; |
| return 0; |
| } |
| } |
| - rtc::Optional<CodecInst> acm_send_codec = audio_coding_->SendCodec(); |
| - if (acm_send_codec) { |
| - codec = *acm_send_codec; |
| - return 0; |
| - } |
| return -1; |
| } |
| @@ -1383,6 +1387,8 @@ int32_t Channel::SetSendCodec(const CodecInst& codec) { |
| } |
| } |
| + cached_send_codec_.reset(); |
| + |
| return 0; |
| } |