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Fix Channel::GetSendCodec when used together with SetEncoder.

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When using the SetEncoder interface, there's no actual CodecInst to return from Channel::GetSendCodec. Before this CL, this was done by calling the ACM, which has functionality for generating a CodecInst with the necessary values even when handed an external encoder. Unfortunately, this call takes a lock and does some extra processing which isn't strictly necessary in this case. Since GetSendCodec is called inside the audio input callback code, this can cause problems.

This CL instead generates a CodecInst in the SetEncoder call and has GetSendCodec simply return that one if it's available. If it isn't the value from codec_manager_ is returned instead, as was the case before injectable audio codec related changes were added to Channel.

BUG=b/38018041

Review-Url: https://codereview.webrtc.org/2924363004
Cr-Commit-Position: refs/heads/master@{#18515}
This commit is contained in:
ossu 2017-06-09 07:30:13 -07:00 committed by Commit Bot
parent 7fdd0676f9
commit 76d29f9bf8
2 changed files with 31 additions and 18 deletions
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47
webrtc/voice_engine/channel.cc
View File

@ -1278,23 +1278,34 @@ bool Channel::SetEncoder(int payload_type,
std::unique_ptr<AudioEncoder> encoder) {
RTC_DCHECK_GE(payload_type, 0);
RTC_DCHECK_LE(payload_type, 127);
// TODO(ossu): Make a CodecInst up for now. It seems like very little of this
// information is actually used, possibly only payload type and clock rate.
CodecInst lies;
lies.pltype = payload_type;
strncpy(lies.plname, "audio", sizeof(lies.plname));
lies.plname[sizeof(lies.plname) - 1] = 0;
// TODO(ossu): Make CodecInsts up, for now: one for the RTP/RTCP module and
// one for for us to keep track of sample rate and number of channels, etc.
// The RTP/RTCP module needs to know the RTP timestamp rate (i.e. clockrate)
// as well as some other things, so we collect this info and send it along.
CodecInst rtp_codec;
rtp_codec.pltype = payload_type;
strncpy(rtp_codec.plname, "audio", sizeof(rtp_codec.plname));
rtp_codec.plname[sizeof(rtp_codec.plname) - 1] = 0;
// Seems unclear if it should be clock rate or sample rate. CodecInst
// 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.channels = encoder->NumChannels();
lies.rate = 0;
rtp_codec.plfreq = encoder->RtpTimestampRateHz();
rtp_codec.pacsize = rtc::CheckedDivExact(
static_cast<int>(encoder->Max10MsFramesInAPacket() * rtp_codec.plfreq),
100);
rtp_codec.channels = encoder->NumChannels();
rtp_codec.rate = 0;
if (_rtpRtcpModule->RegisterSendPayload(lies) != 0) {
// For audio encoding we need, instead, the actual sample rate of the codec.
// The rest of the information should be the same.
CodecInst send_codec = rtp_codec;
send_codec.plfreq = encoder->SampleRateHz();
cached_send_codec_.emplace(send_codec);
if (_rtpRtcpModule->RegisterSendPayload(rtp_codec) != 0) {
_rtpRtcpModule->DeRegisterSendPayload(payload_type);
if (_rtpRtcpModule->RegisterSendPayload(lies) != 0) {
if (_rtpRtcpModule->RegisterSendPayload(rtp_codec) != 0) {
WEBRTC_TRACE(
kTraceError, kTraceVoice, VoEId(_instanceId, _channelId),
"SetEncoder() failed to register codec to RTP/RTCP module");
@ -1343,18 +1354,16 @@ 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 +1392,8 @@ int32_t Channel::SetSendCodec(const CodecInst& codec) {
}
}
cached_send_codec_.reset();
return 0;
}

2
webrtc/voice_engine/channel.h
View File

@ -543,6 +543,8 @@ class Channel
// TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed.
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
rtc::Optional<CodecInst> cached_send_codec_;
rtc::ThreadChecker construction_thread_;
const bool use_twcc_plr_for_ana_;