Refactoring: Use enum for RtpPayloadType.

chrome/renderer/extensions/cast_streaming_native_handler.cc

Index: chrome/renderer/extensions/cast_streaming_native_handler.cc
diff --git a/chrome/renderer/extensions/cast_streaming_native_handler.cc b/chrome/renderer/extensions/cast_streaming_native_handler.cc
index c03f63fc5607ae012e854405569efef0f9a5d0d0..51a1f935e5f3d3a3bffe1ee34e8ce860db65a029 100644
--- a/chrome/renderer/extensions/cast_streaming_native_handler.cc
+++ b/chrome/renderer/extensions/cast_streaming_native_handler.cc
@@ -96,7 +96,11 @@ bool HexDecode(const std::string& input, std::string* output) {
 bool ToCastRtpPayloadParamsOrThrow(v8::Isolate* isolate,
                                    const RtpPayloadParams& ext_params,
                                    CastRtpPayloadParams* cast_params) {
-  cast_params->payload_type = ext_params.payload_type;
+  DCHECK_GE(ext_params.payload_type, media::cast::RTP_PAYLOAD_AUDIO_OPUS);

[miu, 2016/06/20 21:00:27]

The process shouldn't crash on invalid inputs. Instead, the method should return
false.

Suggest you add a RtpPlayloadType::FIRST and LAST to the enum, then just test
the user's input with:

  if (ext_params.payload_type < FIRST || > LAST) {
    isolate->ThrowException(v8::Exception::Error(
        v8::String::NewFromUtf8(isolate, kInvalid...)));
    return false;
  }
  cast_params->payload_type = static_cast<...>(...);

[xjz, 2016/06/21 17:24:31]

Ignored |ext_params.payload_type| and set the |cast_params->payload_type|
according to |codec_name| instead.

+  DCHECK_LE(ext_params.payload_type, media::cast::RTP_PAYLOAD_VIDEO_LAST);
+  cast_params->payload_type =
+      static_cast<media::cast::RtpPayloadType>(ext_params.payload_type);
+
   cast_params->max_latency_ms = ext_params.max_latency;
   cast_params->min_latency_ms =
       ext_params.min_latency ? *ext_params.min_latency : ext_params.max_latency;
@@ -676,23 +680,23 @@ bool CastStreamingNativeHandler::FrameReceiverConfigFromArg(
   if (params->codec_name == "OPUS") {
     config->codec = media::cast::CODEC_AUDIO_OPUS;
     config->rtp_timebase = 48000;
-    config->rtp_payload_type = media::cast::kDefaultRtpAudioPayloadType;
+    config->rtp_payload_type = media::cast::RTP_PAYLOAD_AUDIO_OPUS;
   } else if (params->codec_name == "PCM16") {
     config->codec = media::cast::CODEC_AUDIO_PCM16;
     config->rtp_timebase = 48000;
-    config->rtp_payload_type = media::cast::kDefaultRtpAudioPayloadType;
+    config->rtp_payload_type = media::cast::RTP_PAYLOAD_AUDIO_PCM16;
   } else if (params->codec_name == "AAC") {
     config->codec = media::cast::CODEC_AUDIO_AAC;
     config->rtp_timebase = 48000;
-    config->rtp_payload_type = media::cast::kDefaultRtpAudioPayloadType;
+    config->rtp_payload_type = media::cast::RTP_PAYLOAD_AUDIO_AAC;
   } else if (params->codec_name == "VP8") {
     config->codec = media::cast::CODEC_VIDEO_VP8;
     config->rtp_timebase = 90000;
-    config->rtp_payload_type = media::cast::kDefaultRtpVideoPayloadType;
+    config->rtp_payload_type = media::cast::RTP_PAYLOAD_VIDEO_VP8;
   } else if (params->codec_name == "H264") {
     config->codec = media::cast::CODEC_VIDEO_H264;
     config->rtp_timebase = 90000;
-    config->rtp_payload_type = media::cast::kDefaultRtpVideoPayloadType;
+    config->rtp_payload_type = media::cast::RTP_PAYLOAD_VIDEO_H264;
   }
   if (params->rtp_timebase) {
     config->rtp_timebase = *params->rtp_timebase;