Chromium Code Reviews| Index: webrtc/call/audio_send_stream.cc | 
| diff --git a/webrtc/call/audio_send_stream.cc b/webrtc/call/audio_send_stream.cc | 
| index 6091462470dba0786850e12b889577653706b397..827ecc3aea5f2db7d37c57d17f5265de96bcfddb 100644 | 
| --- a/webrtc/call/audio_send_stream.cc | 
| +++ b/webrtc/call/audio_send_stream.cc | 
| @@ -12,21 +12,6 @@ | 
| #include <string> | 
| -namespace { | 
| - | 
| -std::string ToString(const webrtc::CodecInst& codec_inst) { | 
| - std::stringstream ss; | 
| - ss << "{pltype: " << codec_inst.pltype; | 
| - ss << ", plname: \"" << codec_inst.plname << "\""; | 
| - ss << ", plfreq: " << codec_inst.plfreq; | 
| - ss << ", pacsize: " << codec_inst.pacsize; | 
| - ss << ", channels: " << codec_inst.channels; | 
| - ss << ", rate: " << codec_inst.rate; | 
| - ss << '}'; | 
| - return ss.str(); | 
| -} | 
| -} // namespace | 
| - | 
| namespace webrtc { | 
| AudioSendStream::Stats::Stats() = default; | 
| @@ -44,7 +29,8 @@ std::string AudioSendStream::Config::ToString() const { | 
| ss << ", voe_channel_id: " << voe_channel_id; | 
| ss << ", min_bitrate_bps: " << min_bitrate_bps; | 
| ss << ", max_bitrate_bps: " << max_bitrate_bps; | 
| - ss << ", send_codec_spec: " << send_codec_spec.ToString(); | 
| + ss << ", send_codec_spec: " | 
| + << (send_codec_spec ? send_codec_spec->ToString() : "<unset>"); | 
| ss << '}'; | 
| return ss.str(); | 
| } | 
| @@ -70,24 +56,19 @@ std::string AudioSendStream::Config::Rtp::ToString() const { | 
| return ss.str(); | 
| } | 
| -AudioSendStream::Config::SendCodecSpec::SendCodecSpec() { | 
| - webrtc::CodecInst empty_inst = {0}; | 
| - codec_inst = empty_inst; | 
| - codec_inst.pltype = -1; | 
| -} | 
| +AudioSendStream::Config::SendCodecSpec::SendCodecSpec( | 
| + int payload_type, | 
| 
 
ossu
2017/04/04 15:36:38
git cl format did this.
 
 | 
| + const SdpAudioFormat& format) | 
| + : payload_type(payload_type), format(format) {} | 
| +AudioSendStream::Config::SendCodecSpec::~SendCodecSpec() = default; | 
| std::string AudioSendStream::Config::SendCodecSpec::ToString() const { | 
| std::stringstream ss; | 
| ss << "{nack_enabled: " << (nack_enabled ? "true" : "false"); | 
| ss << ", transport_cc_enabled: " << (transport_cc_enabled ? "true" : "false"); | 
| - ss << ", enable_codec_fec: " << (enable_codec_fec ? "true" : "false"); | 
| - ss << ", enable_opus_dtx: " << (enable_opus_dtx ? "true" : "false"); | 
| - 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 << ", cng_payload_type: " << (cng_payload_type ? *cng_payload_type : -1); | 
| 
 
kwiberg-webrtc
2017/04/06 10:13:29
cng_payload_type.value_or(-1)
 
ossu
2017/04/06 11:14:24
I don't know why I didn't think of that. Thanks!
 
 | 
| + ss << ", payload_type: " << payload_type; | 
| + ss << ", format: " << format; | 
| ss << '}'; | 
| return ss.str(); | 
| } | 
| @@ -96,12 +77,9 @@ bool AudioSendStream::Config::SendCodecSpec::operator==( | 
| const AudioSendStream::Config::SendCodecSpec& rhs) const { | 
| if (nack_enabled == rhs.nack_enabled && | 
| 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) { | 
| + payload_type == rhs.payload_type && format == rhs.format && | 
| + target_bitrate_bps == rhs.target_bitrate_bps) { | 
| return true; | 
| } | 
| return false; |