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Unified Diff: webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h

Issue 1725143003: Changed AudioEncoder::Encode to take an rtc::Buffer* instead of uint8_t* and a maximum size. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Reverted unnecessary change to buffer_unittest.cc Created 4 years, 10 months ago
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Index: webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
index 321dac3567b64bb96ef7db2bbc7ffd369cf3f685..39b1a9f861f122ecaf6bd6d07ccccffa1e370fcb 100644
--- a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
+++ b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
@@ -62,8 +62,7 @@ class AudioEncoderIsacT final : public AudioEncoder {
int GetTargetBitrate() const override;
EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
rtc::ArrayView<const int16_t> audio,
- size_t max_encoded_bytes,
- uint8_t* encoded) override;
+ rtc::Buffer* encoded) override;
void Reset() override;
private:

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