| Index: webrtc/media/engine/webrtcvoiceengine.h
|
| diff --git a/webrtc/media/engine/webrtcvoiceengine.h b/webrtc/media/engine/webrtcvoiceengine.h
|
| index 27ae593cf15e66cc82fc935696075f80ca04883f..edd3e156d134612a1118fbb89c4e88366f07772f 100644
|
| --- a/webrtc/media/engine/webrtcvoiceengine.h
|
| +++ b/webrtc/media/engine/webrtcvoiceengine.h
|
| @@ -37,26 +37,6 @@ class AudioSource;
|
| class VoEWrapper;
|
| class WebRtcVoiceMediaChannel;
|
|
|
| -struct SendCodecSpec {
|
| - SendCodecSpec() {
|
| - webrtc::CodecInst empty_inst = {0};
|
| - codec_inst = empty_inst;
|
| - codec_inst.pltype = -1;
|
| - }
|
| - bool operator==(const SendCodecSpec& rhs) const;
|
| - bool operator!=(const SendCodecSpec& rhs) const;
|
| -
|
| - 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 red_payload_type = -1;
|
| - int cng_payload_type = -1;
|
| - int cng_plfreq = -1;
|
| - webrtc::CodecInst codec_inst;
|
| -};
|
| -
|
| // WebRtcVoiceEngine is a class to be used with CompositeMediaEngine.
|
| // It uses the WebRtc VoiceEngine library for audio handling.
|
| class WebRtcVoiceEngine final : public webrtc::TraceCallback {
|
| @@ -237,8 +217,6 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
|
| bool SetOptions(const AudioOptions& options);
|
| bool SetRecvCodecs(const std::vector<AudioCodec>& codecs);
|
| bool SetSendCodecs(const std::vector<AudioCodec>& codecs);
|
| - bool SetSendCodecs(int channel, const webrtc::RtpParameters& rtp_parameters);
|
| - bool SetSendCodec(int channel, const webrtc::CodecInst& send_codec);
|
| bool SetLocalSource(uint32_t ssrc, AudioSource* source);
|
| bool MuteStream(uint32_t ssrc, bool mute);
|
|
|
| @@ -252,12 +230,6 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
|
| return default_recv_ssrc_ == static_cast<int64_t>(ssrc);
|
| }
|
| bool SetMaxSendBitrate(int bps);
|
| - bool SetChannelSendParameters(int channel,
|
| - const webrtc::RtpParameters& parameters);
|
| - bool SetMaxSendBitrate(int channel, int bps);
|
| - bool HasSendCodec() const {
|
| - return send_codec_spec_.codec_inst.pltype != -1;
|
| - }
|
| bool ValidateRtpParameters(const webrtc::RtpParameters& parameters);
|
| void SetupRecording();
|
|
|
| @@ -295,7 +267,7 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
|
| std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_;
|
| std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
|
|
|
| - SendCodecSpec send_codec_spec_;
|
| + webrtc::AudioSendStream::Config::SendCodecSpec send_codec_spec_;
|
|
|
| RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
|
| };
|
|
|