| Index: webrtc/call/audio_send_stream.h
|
| diff --git a/webrtc/call/audio_send_stream.h b/webrtc/call/audio_send_stream.h
|
| index 42914301911968878d4559a91ac7336c7d622c47..902bec940592cb9b1e6c316c4cc6e90567085f60 100644
|
| --- a/webrtc/call/audio_send_stream.h
|
| +++ b/webrtc/call/audio_send_stream.h
|
| @@ -15,10 +15,11 @@
|
| #include <string>
|
| #include <vector>
|
|
|
| +#include "webrtc/api/audio_codecs/audio_format.h"
|
| #include "webrtc/api/call/transport.h"
|
| #include "webrtc/base/optional.h"
|
| #include "webrtc/config.h"
|
| -#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
|
| +#include "webrtc/modules/audio_coding/codecs/audio_encoder_factory.h"
|
| #include "webrtc/typedefs.h"
|
|
|
| namespace webrtc {
|
| @@ -103,6 +104,7 @@ class AudioSendStream {
|
|
|
| struct SendCodecSpec {
|
| SendCodecSpec();
|
| + ~SendCodecSpec();
|
| std::string ToString() const;
|
|
|
| bool operator==(const SendCodecSpec& rhs) const;
|
| @@ -112,15 +114,14 @@ class AudioSendStream {
|
|
|
| bool nack_enabled = false;
|
| bool transport_cc_enabled = false;
|
| - bool enable_codec_fec = false;
|
| - bool enable_opus_dtx = false;
|
| - int opus_max_playback_rate = 0;
|
| int cng_payload_type = -1;
|
| int cng_plfreq = -1;
|
| - int max_ptime_ms = -1;
|
| - int min_ptime_ms = -1;
|
| - webrtc::CodecInst codec_inst;
|
| + int payload_type;
|
| + rtc::Optional<int> target_bitrate_bps;
|
| + webrtc::AudioCodecSpec format;
|
| } send_codec_spec;
|
| +
|
| + rtc::scoped_refptr<AudioEncoderFactory> encoder_factory;
|
| };
|
|
|
| // Starts stream activity.
|
|
|