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 ef9aa4479b9be95466f25ca24b1a5569c87778df..45e778bf997bde575ff987f17c839ca1ed01d2c7 100644 |
--- a/webrtc/modules/audio_coding/codecs/audio_format_conversion.cc |
+++ b/webrtc/modules/audio_coding/codecs/audio_format_conversion.cc |
@@ -10,11 +10,36 @@ |
#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h" |
+#include <string.h> |
+ |
+#include "webrtc/base/array_view.h" |
#include "webrtc/base/checks.h" |
+#include "webrtc/base/optional.h" |
#include "webrtc/base/safe_conversions.h" |
+#include "webrtc/base/sanitizer.h" |
namespace webrtc { |
+namespace { |
+ |
+CodecInst MakeCodecInst(int payload_type, |
+ const char* name, |
+ int sample_rate, |
+ int 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}; |
+ rtc::MsanMarkUninitialized(rtc::MakeArrayView(&ci, 1)); |
+ ci.pltype = payload_type; |
+ strncpy(ci.plname, name, sizeof(ci.plname)); |
+ ci.plname[sizeof(ci.plname) - 1] = '\0'; |
+ ci.plfreq = sample_rate; |
+ ci.channels = rtc::checked_cast<size_t>(num_channels); |
+ return ci; |
+} |
+ |
+} // namespace |
+ |
SdpAudioFormat CodecInstToSdp(const CodecInst& ci) { |
if (STR_CASE_CMP(ci.plname, "g722") == 0 && ci.plfreq == 16000) { |
RTC_CHECK(ci.channels == 1 || ci.channels == 2); |
@@ -27,4 +52,33 @@ SdpAudioFormat CodecInstToSdp(const CodecInst& ci) { |
} |
} |
+CodecInst SdpToCodecInst(int payload_type, const SdpAudioFormat& audio_format) { |
+ if (STR_CASE_CMP(audio_format.name.c_str(), "g722") == 0 && |
+ audio_format.clockrate_hz == 8000 && |
+ (audio_format.num_channels == 1 || audio_format.num_channels == 2)) { |
+ return MakeCodecInst(payload_type, "g722", 16000, |
+ audio_format.num_channels); |
+ } else if (STR_CASE_CMP(audio_format.name.c_str(), "opus") == 0 && |
+ audio_format.clockrate_hz == 48000 && |
+ audio_format.num_channels == 2) { |
+ const rtc::Optional<int> num_channels = [&] { |
+ const auto stereo = audio_format.parameters.find("stereo"); |
+ if (stereo != audio_format.parameters.end()) { |
+ if (stereo->second == "0") { |
+ return rtc::Optional<int>(1); |
+ } else if (stereo->second == "1") { |
+ return rtc::Optional<int>(2); |
+ } |
+ } |
+ return rtc::Optional<int>(); |
+ }(); |
+ if (num_channels) { |
+ return MakeCodecInst(payload_type, "opus", 48000, *num_channels); |
+ } |
+ } |
+ |
+ return MakeCodecInst(payload_type, audio_format.name.c_str(), |
+ audio_format.clockrate_hz, audio_format.num_channels); |
+} |
+ |
} // namespace webrtc |