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e40468ba3d
Specifically, I'm moving safe_compare.h safe_conversions.h safe_minmax.h They shouldn't be part of the API, and moving them to an appropriate subdirectory of rtc_base/ is a good way to keep track of that. BUG=webrtc:8445 Change-Id: I458531aeb30bcf4291c4bec3bf22a2fffbf054ff Reviewed-on: https://webrtc-review.googlesource.com/20860 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20829}
88 lines
3 KiB
C++
88 lines
3 KiB
C++
/*
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* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_coding/codecs/audio_format_conversion.h"
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#include <string.h>
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#include "api/array_view.h"
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#include "api/optional.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/numerics/safe_conversions.h"
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#include "rtc_base/sanitizer.h"
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namespace webrtc {
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namespace {
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CodecInst MakeCodecInst(int payload_type,
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const char* name,
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int sample_rate,
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size_t num_channels) {
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// Create a CodecInst with some fields set. The remaining fields are zeroed,
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// but we tell MSan to consider them uninitialized.
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CodecInst ci = {0};
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rtc::MsanMarkUninitialized(rtc::MakeArrayView(&ci, 1));
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ci.pltype = payload_type;
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strncpy(ci.plname, name, sizeof(ci.plname));
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ci.plname[sizeof(ci.plname) - 1] = '\0';
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ci.plfreq = sample_rate;
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ci.channels = num_channels;
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return ci;
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}
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} // namespace
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SdpAudioFormat CodecInstToSdp(const CodecInst& ci) {
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if (STR_CASE_CMP(ci.plname, "g722") == 0) {
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RTC_CHECK_EQ(16000, ci.plfreq);
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RTC_CHECK(ci.channels == 1 || ci.channels == 2);
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return {"g722", 8000, ci.channels};
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} else if (STR_CASE_CMP(ci.plname, "opus") == 0) {
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RTC_CHECK_EQ(48000, ci.plfreq);
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RTC_CHECK(ci.channels == 1 || ci.channels == 2);
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return ci.channels == 1
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? SdpAudioFormat("opus", 48000, 2)
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: SdpAudioFormat("opus", 48000, 2, {{"stereo", "1"}});
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} else {
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return {ci.plname, ci.plfreq, ci.channels};
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}
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}
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CodecInst SdpToCodecInst(int payload_type, const SdpAudioFormat& audio_format) {
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if (STR_CASE_CMP(audio_format.name.c_str(), "g722") == 0) {
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RTC_CHECK_EQ(8000, audio_format.clockrate_hz);
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RTC_CHECK(audio_format.num_channels == 1 || audio_format.num_channels == 2);
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return MakeCodecInst(payload_type, "g722", 16000,
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audio_format.num_channels);
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} else if (STR_CASE_CMP(audio_format.name.c_str(), "opus") == 0) {
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RTC_CHECK_EQ(48000, audio_format.clockrate_hz);
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RTC_CHECK_EQ(2, audio_format.num_channels);
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const int num_channels = [&] {
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auto stereo = audio_format.parameters.find("stereo");
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if (stereo != audio_format.parameters.end()) {
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if (stereo->second == "0") {
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return 1;
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} else if (stereo->second == "1") {
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return 2;
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} else {
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RTC_CHECK(false); // Bad stereo parameter.
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}
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}
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return 1; // Default to mono.
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}();
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return MakeCodecInst(payload_type, "opus", 48000, num_channels);
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} else {
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return MakeCodecInst(payload_type, audio_format.name.c_str(),
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audio_format.clockrate_hz, audio_format.num_channels);
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}
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}
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} // namespace webrtc
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