Chromium Code Reviews| Index: webrtc/api/call/audio_send_stream.cc |
| diff --git a/webrtc/api/call/audio_send_stream.cc b/webrtc/api/call/audio_send_stream.cc |
| index 06cbc545d9313846b057ed3432e2862c5c8b9b14..52c30f0987b9efe23ee04821dd2dc9db4e783027 100644 |
| --- a/webrtc/api/call/audio_send_stream.cc |
| +++ b/webrtc/api/call/audio_send_stream.cc |
| @@ -34,6 +34,8 @@ AudioSendStream::Stats::Stats() = default; |
| AudioSendStream::Config::Config(Transport* send_transport) |
| : send_transport(send_transport) {} |
| +AudioSendStream::Config::~Config() = default; |
| + |
| std::string AudioSendStream::Config::ToString() const { |
| std::stringstream ss; |
| ss << "{rtp: " << rtp.ToString(); |
| @@ -82,6 +84,8 @@ std::string AudioSendStream::Config::SendCodecSpec::ToString() const { |
| ss << ", opus_max_playback_rate: " << opus_max_playback_rate; |
| ss << ", cng_payload_type: " << cng_payload_type; |
| ss << ", cng_plfreq: " << cng_plfreq; |
| + ss << ", min_ptime: " << min_ptime_ms; |
| + ss << ", max_ptime: " << max_ptime_ms; |
| ss << ", codec_inst: " << ::ToString(codec_inst); |
| ss << '}'; |
| return ss.str(); |
| @@ -89,30 +93,16 @@ std::string AudioSendStream::Config::SendCodecSpec::ToString() const { |
| bool AudioSendStream::Config::SendCodecSpec::operator==( |
| const AudioSendStream::Config::SendCodecSpec& rhs) const { |
| - if (nack_enabled != rhs.nack_enabled) { |
| - return false; |
| - } |
| - if (transport_cc_enabled != rhs.transport_cc_enabled) { |
| - return false; |
| - } |
| - if (enable_codec_fec != rhs.enable_codec_fec) { |
| - return false; |
| - } |
| - if (enable_opus_dtx != rhs.enable_opus_dtx) { |
| - return false; |
| - } |
| - if (opus_max_playback_rate != rhs.opus_max_playback_rate) { |
| - return false; |
| - } |
| - if (cng_payload_type != rhs.cng_payload_type) { |
| - return false; |
| - } |
| - if (cng_plfreq != rhs.cng_plfreq) { |
| - return false; |
| - } |
| - if (codec_inst != rhs.codec_inst) { |
| - return false; |
| + if (nack_enabled == rhs.nack_enabled && |
|
minyue-webrtc
2016/10/27 14:33:10
refactored per Michael's request
|
| + transport_cc_enabled == rhs.transport_cc_enabled && |
| + enable_codec_fec == rhs.enable_codec_fec && |
| + enable_opus_dtx == rhs.enable_opus_dtx && |
| + opus_max_playback_rate == rhs.opus_max_playback_rate && |
| + cng_payload_type == rhs.cng_payload_type && |
| + cng_plfreq == rhs.cng_plfreq && max_ptime_ms == rhs.max_ptime_ms && |
| + min_ptime_ms == rhs.min_ptime_ms && codec_inst == rhs.codec_inst) { |
| + return true; |
| } |
| - return true; |
| + return false; |
| } |
| } // namespace webrtc |