Index: webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h |
diff --git a/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h b/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h |
index 66f82b22d531039fed2ecc38e2e2b1698c70a637..2587f710aeec9cedb3d593baef3acd284d142c7d 100644 |
--- a/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h |
+++ b/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h |
@@ -43,6 +43,9 @@ class MockAudioEncoder : public AudioEncoder { |
MOCK_METHOD1(SetTargetBitrate, void(int target_bps)); |
MOCK_METHOD1(SetMaxBitrate, void(int max_bps)); |
MOCK_METHOD1(SetMaxPayloadSize, void(int max_payload_size_bytes)); |
+ MOCK_METHOD1(OnReceivedOverhead, void(size_t overhead_bytes_per_packet)); |
+ MOCK_METHOD1(OnReceivedTargetAudioBitrate, |
+ void(int target_audio_bitrate_bps)); |
// Note, we explicitly chose not to create a mock for the Encode method. |
MOCK_METHOD3(EncodeImpl, |
@@ -50,6 +53,10 @@ class MockAudioEncoder : public AudioEncoder { |
rtc::ArrayView<const int16_t> audio, |
rtc::Buffer* encoded)); |
+ void AudioEncoderOnReceivedOverhead(size_t overhead_bytes_per_packet); |
+ |
+ void AudioEncoderOnReceivedTargetAudioBitrate(int target_audio_bitrate_bps); |
+ |
class FakeEncoding { |
public: |
// Creates a functor that will return |info| and adjust the rtc::Buffer |