NetEqImpl::GetDecoderFormat: Return RTP clockrate, not codec sample rate
Well, in fact we need to return both. But return codec sample rate separately and let the SdpAudioFormat contain the RTP clockrate, otherwise we're essentially lying to our callers. Bug: webrtc:11028 Change-Id: I40f36cb9db6b9824404ade6b0515a8312ff97009 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156307 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29444}
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@ -143,6 +143,13 @@ class NetEq {
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enum ReturnCodes { kOK = 0, kFail = -1 };
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// Return type for GetDecoderFormat.
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struct DecoderFormat {
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int sample_rate_hz;
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int num_channels;
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SdpAudioFormat sdp_format;
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};
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// Creates a new NetEq object, with parameters set in |config|. The |config|
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// object will only have to be valid for the duration of the call to this
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// method.
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@ -265,7 +272,7 @@ class NetEq {
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// Returns the decoder info for the given payload type. Returns empty if no
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// such payload type was registered.
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virtual absl::optional<SdpAudioFormat> GetDecoderFormat(
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virtual absl::optional<DecoderFormat> GetDecoderFormat(
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int payload_type) const = 0;
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// Flushes both the packet buffer and the sync buffer.
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@ -392,21 +392,23 @@ int NetEqImpl::last_output_sample_rate_hz() const {
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return last_output_sample_rate_hz_;
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}
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absl::optional<SdpAudioFormat> NetEqImpl::GetDecoderFormat(
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absl::optional<NetEq::DecoderFormat> NetEqImpl::GetDecoderFormat(
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int payload_type) const {
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rtc::CritScope lock(&crit_sect_);
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const DecoderDatabase::DecoderInfo* const di =
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decoder_database_->GetDecoderInfo(payload_type);
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if (!di) {
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return absl::nullopt; // Payload type not registered.
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if (di) {
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const AudioDecoder* const decoder = di->GetDecoder();
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// TODO(kwiberg): Why the special case for RED?
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return DecoderFormat{
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/*sample_rate_hz=*/di->IsRed() ? 8000 : di->SampleRateHz(),
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/*num_channels=*/
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decoder ? rtc::dchecked_cast<int>(decoder->Channels()) : 1,
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/*sdp_format=*/di->GetFormat()};
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} else {
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// Payload type not registered.
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return absl::nullopt;
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}
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SdpAudioFormat format = di->GetFormat();
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// TODO(solenberg): This is legacy but messed up - mixing RTP rate and SR.
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format.clockrate_hz = di->IsRed() ? 8000 : di->SampleRateHz();
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const AudioDecoder* const decoder = di->GetDecoder();
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format.num_channels = decoder ? decoder->Channels() : 1;
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return format;
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}
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void NetEqImpl::FlushBuffers() {
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@ -182,7 +182,7 @@ class NetEqImpl : public webrtc::NetEq {
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int last_output_sample_rate_hz() const override;
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absl::optional<SdpAudioFormat> GetDecoderFormat(
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absl::optional<DecoderFormat> GetDecoderFormat(
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int payload_type) const override;
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// Flushes both the packet buffer and the sync buffer.
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