webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.cc
Danil Chapovalov 0bc58cf876 Replace rtc::Optional with absl::optional in api
This is a no-op change because rtc::Optional is an alias to absl::optional

This CL generated by running script with parameter 'api'
Then undo changes to optional target itself and optional_unittests

find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+

find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;

git cl format

Bug: webrtc:9078
Change-Id: I44093da213369d6a502e33792c694f620f53b779
Reviewed-on: https://webrtc-review.googlesource.com/84621
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23707}
2018-06-21 12:50:03 +00:00

82 lines
2.5 KiB
C++

/*
* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "api/audio_codecs/ilbc/audio_encoder_ilbc.h"
#include <memory>
#include <vector>
#include "common_types.h" // NOLINT(build/include)
#include "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/numerics/safe_minmax.h"
#include "rtc_base/ptr_util.h"
#include "rtc_base/string_to_number.h"
namespace webrtc {
namespace {
int GetIlbcBitrate(int ptime) {
switch (ptime) {
case 20:
case 40:
// 38 bytes per frame of 20 ms => 15200 bits/s.
return 15200;
case 30:
case 60:
// 50 bytes per frame of 30 ms => (approx) 13333 bits/s.
return 13333;
default:
FATAL();
}
}
} // namespace
absl::optional<AudioEncoderIlbcConfig> AudioEncoderIlbc::SdpToConfig(
const SdpAudioFormat& format) {
if (STR_CASE_CMP(format.name.c_str(), "ILBC") != 0 ||
format.clockrate_hz != 8000 || format.num_channels != 1) {
return absl::nullopt;
}
AudioEncoderIlbcConfig config;
auto ptime_iter = format.parameters.find("ptime");
if (ptime_iter != format.parameters.end()) {
auto ptime = rtc::StringToNumber<int>(ptime_iter->second);
if (ptime && *ptime > 0) {
const int whole_packets = *ptime / 10;
config.frame_size_ms = rtc::SafeClamp<int>(whole_packets * 10, 20, 60);
}
}
return config.IsOk() ? absl::optional<AudioEncoderIlbcConfig>(config)
: absl::nullopt;
}
void AudioEncoderIlbc::AppendSupportedEncoders(
std::vector<AudioCodecSpec>* specs) {
const SdpAudioFormat fmt = {"ILBC", 8000, 1};
const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt));
specs->push_back({fmt, info});
}
AudioCodecInfo AudioEncoderIlbc::QueryAudioEncoder(
const AudioEncoderIlbcConfig& config) {
RTC_DCHECK(config.IsOk());
return {8000, 1, GetIlbcBitrate(config.frame_size_ms)};
}
std::unique_ptr<AudioEncoder> AudioEncoderIlbc::MakeAudioEncoder(
const AudioEncoderIlbcConfig& config,
int payload_type,
absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
RTC_DCHECK(config.IsOk());
return rtc::MakeUnique<AudioEncoderIlbcImpl>(config, payload_type);
}
} // namespace webrtc