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Voe::Channel: Turned GetPlayoutFrequency into GetRtpTimestampRateHz.

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This gets rid of a bit of codec-specific code in VoE.

BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/2355483003
Cr-Commit-Position: refs/heads/master@{#14614}
This commit is contained in:
ossu
2016-10-12 11:04:10 -07:00
committed by Commit bot
parent 872f614111
commit e280cdeb74
6 changed files with 42 additions and 22 deletions
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9
webrtc/modules/audio_coding/acm2/acm_receiver.cc
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@ -98,6 +98,8 @@ int AcmReceiver::InsertPacket(const WebRtcRTPHeader& rtp_header,
}
} else {
last_audio_decoder_ = ci;
last_audio_format_ = neteq_->GetDecoderFormat(ci->pltype);
RTC_DCHECK(last_audio_format_);
last_packet_sample_rate_hz_ = rtc::Optional<int>(ci->plfreq);
}
@ -263,6 +265,7 @@ void AcmReceiver::RemoveAllCodecs() {
rtc::CritScope lock(&crit_sect_);
neteq_->RemoveAllPayloadTypes();
last_audio_decoder_ = rtc::Optional<CodecInst>();
last_audio_format_ = rtc::Optional<SdpAudioFormat>();
last_packet_sample_rate_hz_ = rtc::Optional<int>();
}
@ -275,6 +278,7 @@ int AcmReceiver::RemoveCodec(uint8_t payload_type) {
}
if (last_audio_decoder_ && payload_type == last_audio_decoder_->pltype) {
last_audio_decoder_ = rtc::Optional<CodecInst>();
last_audio_format_ = rtc::Optional<SdpAudioFormat>();
last_packet_sample_rate_hz_ = rtc::Optional<int>();
}
return 0;
@ -297,6 +301,11 @@ int AcmReceiver::LastAudioCodec(CodecInst* codec) const {
return 0;
}
rtc::Optional<SdpAudioFormat> AcmReceiver::LastAudioFormat() const {
rtc::CritScope lock(&crit_sect_);
return last_audio_format_;
}
void AcmReceiver::GetNetworkStatistics(NetworkStatistics* acm_stat) {
NetEqNetworkStatistics neteq_stat;
// NetEq function always returns zero, so we don't check the return value.