Index: webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h |
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h |
index 150a84162cd80af34f7f367e7c7b4fdb83cdf97c..f0829e63ff491aaa1692bfeca1ab75e891afd97b 100644 |
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h |
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h |
@@ -22,6 +22,10 @@ |
namespace webrtc { |
+namespace { |
+class PacketLossFractionSmoother; |
+} |
kwiberg-webrtc
2016/10/11 14:56:19
Style guide says "Don't use unnamed namespaces in
minyue-webrtc
2016/10/11 15:31:09
Thanks! It reads better.
|
+ |
struct CodecInst; |
class AudioEncoderOpus final : public AudioEncoder { |
@@ -49,6 +53,7 @@ class AudioEncoderOpus final : public AudioEncoder { |
int max_playback_rate_hz = 48000; |
int complexity = kDefaultComplexity; |
bool dtx_enabled = false; |
+ const Clock* clock = nullptr; |
private: |
#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_ARCH_ARM) |
@@ -133,6 +138,7 @@ class AudioEncoderOpus final : public AudioEncoder { |
uint32_t first_timestamp_in_buffer_; |
size_t num_channels_to_encode_; |
int next_frame_length_ms_; |
+ std::unique_ptr<PacketLossFractionSmoother> packet_loss_fraction_smoother_; |
AudioNetworkAdaptorCreator audio_network_adaptor_creator_; |
std::unique_ptr<AudioNetworkAdaptor> audio_network_adaptor_; |