Explicitly use the Opus DTX encoder state.
Use the DTX state from inside the Opus encoder instead of trying to mimic the logic outside. Bug: None Change-Id: I852044fee261a5b7f9255c557a27adfd0b1701bd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362640 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Lionel Koenig <lionelk@webrtc.org> Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43034}
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@ -378,8 +378,7 @@ AudioEncoderOpusImpl::AudioEncoderOpusImpl(
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inst_(nullptr),
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packet_loss_fraction_smoother_(new PacketLossFractionSmoother()),
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audio_network_adaptor_creator_(audio_network_adaptor_creator),
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bitrate_smoother_(std::move(bitrate_smoother)),
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consecutive_dtx_frames_(0) {
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bitrate_smoother_(std::move(bitrate_smoother)) {
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RTC_DCHECK(0 <= payload_type && payload_type <= 127);
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// Sanity check of the redundant payload type field that we want to get rid
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@ -387,6 +386,7 @@ AudioEncoderOpusImpl::AudioEncoderOpusImpl(
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RTC_CHECK(config.payload_type == -1 || config.payload_type == payload_type);
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RTC_CHECK(RecreateEncoderInstance(config));
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SetProjectedPacketLossRate(packet_loss_rate_);
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}
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@ -609,8 +609,6 @@ AudioEncoder::EncodedInfo AudioEncoderOpusImpl::EncodeImpl(
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});
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input_buffer_.clear();
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bool dtx_frame = (info.encoded_bytes <= 2);
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// Will use new packet size for next encoding.
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config_.frame_size_ms = next_frame_length_ms_;
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@ -625,15 +623,9 @@ AudioEncoder::EncodedInfo AudioEncoderOpusImpl::EncodeImpl(
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info.encoded_timestamp = first_timestamp_in_buffer_;
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info.payload_type = payload_type_;
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info.send_even_if_empty = true; // Allows Opus to send empty packets.
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// After 20 DTX frames (MAX_CONSECUTIVE_DTX) Opus will send a frame
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// coding the background noise. Avoid flagging this frame as speech
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// (even though there is a probability of the frame being speech).
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info.speech = !dtx_frame && (consecutive_dtx_frames_ != 20);
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info.speech = WebRtcOpus_GetInDtx(inst_) == 0;
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info.encoder_type = CodecType::kOpus;
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// Increase or reset DTX counter.
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consecutive_dtx_frames_ = (dtx_frame) ? (consecutive_dtx_frames_ + 1) : (0);
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return info;
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}
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@ -179,7 +179,6 @@ class AudioEncoderOpusImpl final : public AudioEncoder {
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std::optional<size_t> overhead_bytes_per_packet_;
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const std::unique_ptr<SmoothingFilter> bitrate_smoother_;
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std::optional<int64_t> bitrate_smoother_last_update_time_;
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int consecutive_dtx_frames_;
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friend struct AudioEncoderOpus;
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};
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@ -310,6 +310,16 @@ int16_t WebRtcOpus_GetUseDtx(OpusEncInst* inst) {
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return -1;
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}
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int16_t WebRtcOpus_GetInDtx(OpusEncInst* inst) {
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if (inst) {
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opus_int32 in_dtx;
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if (ENCODER_CTL(inst, OPUS_GET_IN_DTX(&in_dtx)) == 0) {
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return in_dtx;
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}
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}
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return -1;
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}
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int16_t WebRtcOpus_EnableCbr(OpusEncInst* inst) {
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if (inst) {
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return ENCODER_CTL(inst, OPUS_SET_VBR(0));
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@ -245,6 +245,20 @@ int16_t WebRtcOpus_DisableDtx(OpusEncInst* inst);
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*/
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int16_t WebRtcOpus_GetUseDtx(OpusEncInst* inst);
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/****************************************************************************
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* WebRtcOpus_GetUseDtx()
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*
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* This function gets if the encoder is in DTX.
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*
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* Input:
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* - inst : Encoder context
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*
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* Return value : 0 - Encoder is not DTX.
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* 1 - Encoder is in DTX.
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* -1 - Error.
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*/
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int16_t WebRtcOpus_GetInDtx(OpusEncInst* inst);
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/****************************************************************************
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* WebRtcOpus_EnableCbr()
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*
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