Voe::Channel: Turned GetPlayoutFrequency into GetRtpTimestampRateHz.
This gets rid of a bit of codec-specific code in VoE.
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2355483003
Cr-Original-Commit-Position: refs/heads/master@{#14614}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: e280cdeb744c2e0c40fa1da24dce59436eda2e39
diff --git a/modules/audio_coding/acm2/acm_receiver.cc b/modules/audio_coding/acm2/acm_receiver.cc
index 0b19310..73518b8 100644
--- a/modules/audio_coding/acm2/acm_receiver.cc
+++ b/modules/audio_coding/acm2/acm_receiver.cc
@@ -98,6 +98,8 @@
}
} 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 @@
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 @@
}
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 @@
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.
diff --git a/modules/audio_coding/acm2/acm_receiver.h b/modules/audio_coding/acm2/acm_receiver.h
index a9550fb..6415074 100644
--- a/modules/audio_coding/acm2/acm_receiver.h
+++ b/modules/audio_coding/acm2/acm_receiver.h
@@ -209,6 +209,8 @@
//
int LastAudioCodec(CodecInst* codec) const;
+ rtc::Optional<SdpAudioFormat> LastAudioFormat() const;
+
//
// Get a decoder given its registered payload-type.
//
@@ -273,6 +275,7 @@
rtc::CriticalSection crit_sect_;
rtc::Optional<CodecInst> last_audio_decoder_ GUARDED_BY(crit_sect_);
+ rtc::Optional<SdpAudioFormat> last_audio_format_ GUARDED_BY(crit_sect_);
ACMResampler resampler_ GUARDED_BY(crit_sect_);
std::unique_ptr<int16_t[]> last_audio_buffer_ GUARDED_BY(crit_sect_);
CallStatistics call_stats_ GUARDED_BY(crit_sect_);
diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc
index a845c01..ee1034b 100644
--- a/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/acm2/audio_coding_module.cc
@@ -138,6 +138,8 @@
// Get current received codec.
int ReceiveCodec(CodecInst* current_codec) const override;
+ rtc::Optional<SdpAudioFormat> ReceiveFormat() const override;
+
// Incoming packet from network parsed and ready for decode.
int IncomingPacket(const uint8_t* incoming_payload,
const size_t payload_length,
@@ -1087,6 +1089,11 @@
return receiver_.LastAudioCodec(current_codec);
}
+rtc::Optional<SdpAudioFormat> AudioCodingModuleImpl::ReceiveFormat() const {
+ rtc::CritScope lock(&acm_crit_sect_);
+ return receiver_.LastAudioFormat();
+}
+
// Incoming packet from network parsed and ready for decode.
int AudioCodingModuleImpl::IncomingPacket(const uint8_t* incoming_payload,
const size_t payload_length,
diff --git a/modules/audio_coding/include/audio_coding_module.h b/modules/audio_coding/include/audio_coding_module.h
index e62bbd7..a0dafec 100644
--- a/modules/audio_coding/include/audio_coding_module.h
+++ b/modules/audio_coding/include/audio_coding_module.h
@@ -553,6 +553,17 @@
virtual int32_t ReceiveCodec(CodecInst* curr_receive_codec) const = 0;
///////////////////////////////////////////////////////////////////////////
+ // rtc::Optional<SdpAudioFormat> ReceiveFormat()
+ // Get the format associated with last received payload.
+ //
+ // Return value:
+ // An SdpAudioFormat describing the format associated with the last
+ // received payload.
+ // An empty Optional if no payload has yet been received.
+ //
+ virtual rtc::Optional<SdpAudioFormat> ReceiveFormat() const = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
// int32_t IncomingPacket()
// Call this function to insert a parsed RTP packet into ACM.
//
diff --git a/voice_engine/channel.cc b/voice_engine/channel.cc
index 0965963..6dcef5f 100644
--- a/voice_engine/channel.cc
+++ b/voice_engine/channel.cc
@@ -718,7 +718,7 @@
rtp_ts_wraparound_handler_->Unwrap(audioFrame->timestamp_);
audioFrame->elapsed_time_ms_ =
(unwrap_timestamp - capture_start_rtp_time_stamp_) /
- (GetPlayoutFrequency() / 1000);
+ (GetRtpTimestampRateHz() / 1000);
{
rtc::CritScope lock(&ts_stats_lock_);
@@ -3162,7 +3162,7 @@
uint32_t playout_timestamp = *jitter_buffer_playout_timestamp_;
// Remove the playout delay.
- playout_timestamp -= (delay_ms * (GetPlayoutFrequency() / 1000));
+ playout_timestamp -= (delay_ms * (GetRtpTimestampRateHz() / 1000));
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::UpdatePlayoutTimestamp() => playoutTimestamp = %lu",
@@ -3222,25 +3222,15 @@
return error;
}
-int32_t Channel::GetPlayoutFrequency() const {
- int32_t playout_frequency = audio_coding_->PlayoutFrequency();
- CodecInst current_recive_codec;
- if (audio_coding_->ReceiveCodec(¤t_recive_codec) == 0) {
- if (STR_CASE_CMP("G722", current_recive_codec.plname) == 0) {
- // Even though the actual sampling rate for G.722 audio is
- // 16,000 Hz, the RTP clock rate for the G722 payload format is
- // 8,000 Hz because that value was erroneously assigned in
- // RFC 1890 and must remain unchanged for backward compatibility.
- playout_frequency = 8000;
- } else if (STR_CASE_CMP("opus", current_recive_codec.plname) == 0) {
- // We are resampling Opus internally to 32,000 Hz until all our
- // DSP routines can operate at 48,000 Hz, but the RTP clock
- // rate for the Opus payload format is standardized to 48,000 Hz,
- // because that is the maximum supported decoding sampling rate.
- playout_frequency = 48000;
- }
- }
- return playout_frequency;
+int Channel::GetRtpTimestampRateHz() const {
+ const auto format = audio_coding_->ReceiveFormat();
+ // Default to the playout frequency if we've not gotten any packets yet.
+ // TODO(ossu): Zero clockrate can only happen if we've added an external
+ // decoder for a format we don't support internally. Remove once that way of
+ // adding decoders is gone!
+ return (format && format->clockrate_hz != 0)
+ ? format->clockrate_hz
+ : audio_coding_->PlayoutFrequency();
}
int64_t Channel::GetRTT(bool allow_associate_channel) const {
diff --git a/voice_engine/channel.h b/voice_engine/channel.h
index fc123e3..bf279d8 100644
--- a/voice_engine/channel.h
+++ b/voice_engine/channel.h
@@ -446,7 +446,7 @@
RTPExtensionType type,
unsigned char id);
- int32_t GetPlayoutFrequency() const;
+ int GetRtpTimestampRateHz() const;
int64_t GetRTT(bool allow_associate_channel) const;
rtc::CriticalSection _fileCritSect;
