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Unified Diff: webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc

Issue 1881003003: Reland Remove the deprecated EncodeInternal interface from AudioEncoder (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Renamed ApproximateEncodedBytes to SufficientOutputBufferSize in Opus Created 4 years, 8 months ago
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Index: webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc
diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc
index 254c2f420bf097c333527f82285094594bfc7c45..f00b2432f2430a47bfb67be58c276539107f9a53 100644
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc
@@ -104,7 +104,6 @@ void ConvertEncodedInfoToFragmentationHeader(
class RawAudioEncoderWrapper final : public AudioEncoder {
public:
RawAudioEncoderWrapper(AudioEncoder* enc) : enc_(enc) {}
- size_t MaxEncodedBytes() const override { return enc_->MaxEncodedBytes(); }
int SampleRateHz() const override { return enc_->SampleRateHz(); }
size_t NumChannels() const override { return enc_->NumChannels(); }
int RtpTimestampRateHz() const override { return enc_->RtpTimestampRateHz(); }
@@ -120,13 +119,6 @@ class RawAudioEncoderWrapper final : public AudioEncoder {
rtc::Buffer* encoded) override {
return enc_->Encode(rtp_timestamp, audio, encoded);
}
- EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
- rtc::ArrayView<const int16_t> audio,
- size_t max_encoded_bytes,
- uint8_t* encoded) override {
- return enc_->EncodeInternal(rtp_timestamp, audio, max_encoded_bytes,
- encoded);
- }
void Reset() override { return enc_->Reset(); }
bool SetFec(bool enable) override { return enc_->SetFec(enable); }
bool SetDtx(bool enable) override { return enc_->SetDtx(enable); }
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