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The following APIs on AudioCodingModule are deprecated with this CL: static int NumberOfCodecs(); static int Codec(int, CodecInst*); static int Codec(const char*, CodecInst*, int, size_t); static int Codec(const char*, int, size_t); absl::optional<CodecInst> SendCodec() const; bool RegisterReceiveCodec(int, const SdpAudioFormat&); int RegisterExternalReceiveCodec(int, AudioDecoder*, int, int, const std::string&); int UnregisterReceiveCodec(uint8_t); int32_t ReceiveCodec(CodecInst*); absl::optional<SdpAudioFormat> ReceiveFormat(); As well as this method on RtpRtcp module: int32_t RegisterSendPayload(const CodecInst&); Bug: webrtc:7626 Change-Id: I1230732136f1fe9048cf74afdeab767ca57ac9ce Reviewed-on: https://webrtc-review.googlesource.com/c/113816 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26025} |
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| .. | ||
| acm_receive_test.cc | ||
| acm_receive_test.h | ||
| acm_receiver.cc | ||
| acm_receiver.h | ||
| acm_receiver_unittest.cc | ||
| acm_resampler.cc | ||
| acm_resampler.h | ||
| acm_send_test.cc | ||
| acm_send_test.h | ||
| audio_coding_module.cc | ||
| audio_coding_module_unittest.cc | ||
| call_statistics.cc | ||
| call_statistics.h | ||
| call_statistics_unittest.cc | ||