Index: webrtc/media/engine/webrtcvoiceengine.h |
diff --git a/webrtc/media/engine/webrtcvoiceengine.h b/webrtc/media/engine/webrtcvoiceengine.h |
index 71bd874347d9103547adb854a49de98457f53db6..b675284e55849a0c1a6242d0168083a148a1419e 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); |
@@ -251,12 +229,6 @@ 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(); |
@@ -293,7 +265,7 @@ 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); |
}; |