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Delete use of STR_CASE_CMP, replaced with absl::EqualsIgnoreCase.

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Bug: webrtc:5876
Change-Id: Ica2d47ca45b8ef01a548d8dbe31dbed740a0ebda
Reviewed-on: https://webrtc-review.googlesource.com/c/106820
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25306}
This commit is contained in:
Niels Möller
2018-10-22 09:48:08 +02:00
committed by Commit Bot
parent 01cf44d397
commit 2edab4c026
42 changed files with 146 additions and 142 deletions
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9
modules/audio_coding/codecs/audio_format_conversion.cc
View File

@ -12,6 +12,7 @@
#include <string.h>
#include "absl/strings/match.h"
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "rtc_base/checks.h"
@ -41,11 +42,11 @@ CodecInst MakeCodecInst(int payload_type,
} // namespace
SdpAudioFormat CodecInstToSdp(const CodecInst& ci) {
if (STR_CASE_CMP(ci.plname, "g722") == 0) {
if (absl::EqualsIgnoreCase(ci.plname, "g722")) {
RTC_CHECK_EQ(16000, ci.plfreq);
RTC_CHECK(ci.channels == 1 || ci.channels == 2);
return {"g722", 8000, ci.channels};
} else if (STR_CASE_CMP(ci.plname, "opus") == 0) {
} else if (absl::EqualsIgnoreCase(ci.plname, "opus")) {
RTC_CHECK_EQ(48000, ci.plfreq);
RTC_CHECK(ci.channels == 1 || ci.channels == 2);
return ci.channels == 1
@ -57,12 +58,12 @@ 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) {
if (absl::EqualsIgnoreCase(audio_format.name, "g722")) {
RTC_CHECK_EQ(8000, audio_format.clockrate_hz);
RTC_CHECK(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) {
} else if (absl::EqualsIgnoreCase(audio_format.name, "opus")) {
RTC_CHECK_EQ(48000, audio_format.clockrate_hz);
RTC_CHECK_EQ(2, audio_format.num_channels);
const int num_channels = [&] {

6
modules/audio_coding/codecs/opus/audio_encoder_opus.cc
View File

@ -15,7 +15,7 @@
#include <utility>
#include "absl/memory/memory.h"
#include "common_types.h" // NOLINT(build/include)
#include "absl/strings/match.h"
#include "modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
#include "modules/audio_coding/audio_network_adaptor/controller_manager.h"
#include "modules/audio_coding/codecs/opus/opus_interface.h"
@ -316,7 +316,7 @@ std::unique_ptr<AudioEncoder> AudioEncoderOpusImpl::MakeAudioEncoder(
absl::optional<AudioCodecInfo> AudioEncoderOpusImpl::QueryAudioEncoder(
const SdpAudioFormat& format) {
if (STR_CASE_CMP(format.name.c_str(), GetPayloadName()) == 0 &&
if (absl::EqualsIgnoreCase(format.name, GetPayloadName()) &&
format.clockrate_hz == 48000 && format.num_channels == 2) {
const size_t num_channels = GetChannelCount(format);
const int bitrate =
@ -348,7 +348,7 @@ AudioEncoderOpusConfig AudioEncoderOpusImpl::CreateConfig(
absl::optional<AudioEncoderOpusConfig> AudioEncoderOpusImpl::SdpToConfig(
const SdpAudioFormat& format) {
if (STR_CASE_CMP(format.name.c_str(), "opus") != 0 ||
if (!absl::EqualsIgnoreCase(format.name, "opus") ||
format.clockrate_hz != 48000 || format.num_channels != 2) {
return absl::nullopt;
}