Use Abseil container algorithms in media/
Bug: None
Change-Id: I292e3401bbf19a66271dd5ef2b3ca4f8dcfd155d
Reviewed-on: https://webrtc-review.googlesource.com/c/120003
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26434}diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index 8089a0f..67d89fd 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -19,6 +19,7 @@
#include <utility>
#include <vector>
+#include "absl/algorithm/container.h"
#include "absl/strings/match.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/call/audio_sink.h"
@@ -125,12 +126,10 @@
return true;
}
std::vector<int> payload_types;
- for (const AudioCodec& codec : codecs) {
- payload_types.push_back(codec.id);
- }
- std::sort(payload_types.begin(), payload_types.end());
- auto it = std::unique(payload_types.begin(), payload_types.end());
- return it == payload_types.end();
+ absl::c_transform(codecs, std::back_inserter(payload_types),
+ [](const AudioCodec& codec) { return codec.id; });
+ absl::c_sort(payload_types);
+ return absl::c_adjacent_find(payload_types) == payload_types.end();
}
absl::optional<std::string> GetAudioNetworkAdaptorConfig(
@@ -579,7 +578,7 @@
void WebRtcVoiceEngine::UnregisterChannel(WebRtcVoiceMediaChannel* channel) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
- auto it = std::find(channels_.begin(), channels_.end(), channel);
+ auto it = absl::c_find(channels_, channel);
RTC_DCHECK(it != channels_.end());
channels_.erase(it);
}
@@ -2029,9 +2028,7 @@
if (!GetRtpSsrc(packet->cdata(), packet->size(), &ssrc)) {
return;
}
- RTC_DCHECK(std::find(unsignaled_recv_ssrcs_.begin(),
- unsignaled_recv_ssrcs_.end(),
- ssrc) == unsignaled_recv_ssrcs_.end());
+ RTC_DCHECK(!absl::c_linear_search(unsignaled_recv_ssrcs_, ssrc));
// Add new stream.
StreamParams sp = unsignaled_stream_params_;
@@ -2181,7 +2178,7 @@
// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=8158
if (!unsignaled_recv_ssrcs_.empty()) {
auto end_it = --unsignaled_recv_ssrcs_.end();
- if (std::find(unsignaled_recv_ssrcs_.begin(), end_it, ssrc) != end_it) {
+ if (absl::linear_search(unsignaled_recv_ssrcs_.begin(), end_it, ssrc)) {
continue;
}
}
@@ -2280,8 +2277,7 @@
bool WebRtcVoiceMediaChannel::MaybeDeregisterUnsignaledRecvStream(
uint32_t ssrc) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
- auto it = std::find(unsignaled_recv_ssrcs_.begin(),
- unsignaled_recv_ssrcs_.end(), ssrc);
+ auto it = absl::c_find(unsignaled_recv_ssrcs_, ssrc);
if (it != unsignaled_recv_ssrcs_.end()) {
unsignaled_recv_ssrcs_.erase(it);
return true;