mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-15 14:50:39 +01:00
aefed55c25
First batch of applying iwyu to the repo. Done with: > ./tools_webrtc/iwyu/apply-iwyu api > git add api/[a-s]* > python3 gn_autodeps.py ~/local/webrtc/src out/Default Last step is a custom script I wrote to automatically apply new required dependencies for target in gn, which saved tons of time manually going over the files and fixing. If this is something that interest others, I can submit it as well. Bug: webrtc:42226242 Change-Id: Id109e77f50835827495bc4512880c4ec9ae175f6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343680 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Dor Hen <dorhen@meta.com> Cr-Commit-Position: refs/heads/main@{#42512}
87 lines
2.8 KiB
C++
87 lines
2.8 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/L16/audio_encoder_L16.h"
|
|
|
|
#include <stddef.h>
|
|
|
|
#include <map>
|
|
#include <memory>
|
|
#include <utility>
|
|
#include <vector>
|
|
|
|
#include "absl/strings/match.h"
|
|
#include "absl/types/optional.h"
|
|
#include "api/audio_codecs/audio_codec_pair_id.h"
|
|
#include "api/audio_codecs/audio_encoder.h"
|
|
#include "api/audio_codecs/audio_format.h"
|
|
#include "api/field_trials_view.h"
|
|
#include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
|
|
#include "modules/audio_coding/codecs/pcm16b/pcm16b_common.h"
|
|
#include "rtc_base/checks.h"
|
|
#include "rtc_base/numerics/safe_conversions.h"
|
|
#include "rtc_base/numerics/safe_minmax.h"
|
|
#include "rtc_base/string_to_number.h"
|
|
|
|
namespace webrtc {
|
|
|
|
absl::optional<AudioEncoderL16::Config> AudioEncoderL16::SdpToConfig(
|
|
const SdpAudioFormat& format) {
|
|
if (!rtc::IsValueInRangeForNumericType<int>(format.num_channels)) {
|
|
RTC_DCHECK_NOTREACHED();
|
|
return absl::nullopt;
|
|
}
|
|
Config config;
|
|
config.sample_rate_hz = format.clockrate_hz;
|
|
config.num_channels = rtc::dchecked_cast<int>(format.num_channels);
|
|
auto ptime_iter = format.parameters.find("ptime");
|
|
if (ptime_iter != format.parameters.end()) {
|
|
const auto ptime = rtc::StringToNumber<int>(ptime_iter->second);
|
|
if (ptime && *ptime > 0) {
|
|
config.frame_size_ms = rtc::SafeClamp(10 * (*ptime / 10), 10, 60);
|
|
}
|
|
}
|
|
if (absl::EqualsIgnoreCase(format.name, "L16") && config.IsOk()) {
|
|
return config;
|
|
}
|
|
return absl::nullopt;
|
|
}
|
|
|
|
void AudioEncoderL16::AppendSupportedEncoders(
|
|
std::vector<AudioCodecSpec>* specs) {
|
|
Pcm16BAppendSupportedCodecSpecs(specs);
|
|
}
|
|
|
|
AudioCodecInfo AudioEncoderL16::QueryAudioEncoder(
|
|
const AudioEncoderL16::Config& config) {
|
|
RTC_DCHECK(config.IsOk());
|
|
return {config.sample_rate_hz,
|
|
rtc::dchecked_cast<size_t>(config.num_channels),
|
|
config.sample_rate_hz * config.num_channels * 16};
|
|
}
|
|
|
|
std::unique_ptr<AudioEncoder> AudioEncoderL16::MakeAudioEncoder(
|
|
const AudioEncoderL16::Config& config,
|
|
int payload_type,
|
|
absl::optional<AudioCodecPairId> /*codec_pair_id*/,
|
|
const FieldTrialsView* field_trials) {
|
|
AudioEncoderPcm16B::Config c;
|
|
c.sample_rate_hz = config.sample_rate_hz;
|
|
c.num_channels = config.num_channels;
|
|
c.frame_size_ms = config.frame_size_ms;
|
|
c.payload_type = payload_type;
|
|
if (!config.IsOk()) {
|
|
RTC_DCHECK_NOTREACHED();
|
|
return nullptr;
|
|
}
|
|
return std::make_unique<AudioEncoderPcm16B>(c);
|
|
}
|
|
|
|
} // namespace webrtc
|