| Index: webrtc/modules/audio_coding/codecs/audio_format_conversion.cc
|
| diff --git a/webrtc/modules/audio_coding/codecs/audio_format_conversion.cc b/webrtc/modules/audio_coding/codecs/audio_format_conversion.cc
|
| index 5a69ae431d1c3e4ead6e9ffd214a85efffc27455..a858053c78010d30cb8df47a79141dad8f1bd175 100644
|
| --- a/webrtc/modules/audio_coding/codecs/audio_format_conversion.cc
|
| +++ b/webrtc/modules/audio_coding/codecs/audio_format_conversion.cc
|
| @@ -25,7 +25,7 @@ namespace {
|
| CodecInst MakeCodecInst(int payload_type,
|
| const char* name,
|
| int sample_rate,
|
| - int num_channels) {
|
| + size_t num_channels) {
|
| // Create a CodecInst with some fields set. The remaining fields are zeroed,
|
| // but we tell MSan to consider them uninitialized.
|
| CodecInst ci = {0};
|
| @@ -34,7 +34,7 @@ CodecInst MakeCodecInst(int payload_type,
|
| strncpy(ci.plname, name, sizeof(ci.plname));
|
| ci.plname[sizeof(ci.plname) - 1] = '\0';
|
| ci.plfreq = sample_rate;
|
| - ci.channels = rtc::dchecked_cast<size_t>(num_channels);
|
| + ci.channels = num_channels;
|
| return ci;
|
| }
|
|
|
| @@ -44,7 +44,7 @@ SdpAudioFormat CodecInstToSdp(const CodecInst& ci) {
|
| if (STR_CASE_CMP(ci.plname, "g722") == 0) {
|
| RTC_CHECK_EQ(16000, ci.plfreq);
|
| RTC_CHECK(ci.channels == 1 || ci.channels == 2);
|
| - return {"g722", 8000, static_cast<int>(ci.channels)};
|
| + return {"g722", 8000, ci.channels};
|
| } else if (STR_CASE_CMP(ci.plname, "opus") == 0) {
|
| RTC_CHECK_EQ(48000, ci.plfreq);
|
| RTC_CHECK(ci.channels == 1 || ci.channels == 2);
|
| @@ -52,7 +52,7 @@ SdpAudioFormat CodecInstToSdp(const CodecInst& ci) {
|
| ? SdpAudioFormat("opus", 48000, 2)
|
| : SdpAudioFormat("opus", 48000, 2, {{"stereo", "1"}});
|
| } else {
|
| - return {ci.plname, ci.plfreq, rtc::checked_cast<int>(ci.channels)};
|
| + return {ci.plname, ci.plfreq, ci.channels};
|
| }
|
| }
|
|
|
|
|