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a4d873786f
This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505}
88 lines
3 KiB
C++
88 lines
3 KiB
C++
/*
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* Copyright (c) 2017 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 "api/audio_codecs/g711/audio_encoder_g711.h"
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#include <memory>
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#include <vector>
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#include "absl/memory/memory.h"
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#include "absl/strings/match.h"
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#include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
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#include "rtc_base/numerics/safe_conversions.h"
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#include "rtc_base/numerics/safe_minmax.h"
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#include "rtc_base/string_to_number.h"
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namespace webrtc {
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absl::optional<AudioEncoderG711::Config> AudioEncoderG711::SdpToConfig(
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const SdpAudioFormat& format) {
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const bool is_pcmu = absl::EqualsIgnoreCase(format.name, "PCMU");
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const bool is_pcma = absl::EqualsIgnoreCase(format.name, "PCMA");
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if (format.clockrate_hz == 8000 && format.num_channels >= 1 &&
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(is_pcmu || is_pcma)) {
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Config config;
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config.type = is_pcmu ? Config::Type::kPcmU : Config::Type::kPcmA;
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config.num_channels = rtc::dchecked_cast<int>(format.num_channels);
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config.frame_size_ms = 20;
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auto ptime_iter = format.parameters.find("ptime");
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if (ptime_iter != format.parameters.end()) {
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const auto ptime = rtc::StringToNumber<int>(ptime_iter->second);
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if (ptime && *ptime > 0) {
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config.frame_size_ms = rtc::SafeClamp(10 * (*ptime / 10), 10, 60);
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}
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}
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RTC_DCHECK(config.IsOk());
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return config;
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} else {
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return absl::nullopt;
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}
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}
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void AudioEncoderG711::AppendSupportedEncoders(
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std::vector<AudioCodecSpec>* specs) {
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for (const char* type : {"PCMU", "PCMA"}) {
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specs->push_back({{type, 8000, 1}, {8000, 1, 64000}});
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}
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}
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AudioCodecInfo AudioEncoderG711::QueryAudioEncoder(const Config& config) {
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RTC_DCHECK(config.IsOk());
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return {8000, rtc::dchecked_cast<size_t>(config.num_channels),
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64000 * config.num_channels};
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}
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std::unique_ptr<AudioEncoder> AudioEncoderG711::MakeAudioEncoder(
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const Config& config,
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int payload_type,
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absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
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RTC_DCHECK(config.IsOk());
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switch (config.type) {
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case Config::Type::kPcmU: {
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AudioEncoderPcmU::Config impl_config;
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impl_config.num_channels = config.num_channels;
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impl_config.frame_size_ms = config.frame_size_ms;
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impl_config.payload_type = payload_type;
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return absl::make_unique<AudioEncoderPcmU>(impl_config);
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}
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case Config::Type::kPcmA: {
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AudioEncoderPcmA::Config impl_config;
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impl_config.num_channels = config.num_channels;
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impl_config.frame_size_ms = config.frame_size_ms;
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impl_config.payload_type = payload_type;
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return absl::make_unique<AudioEncoderPcmA>(impl_config);
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}
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default: {
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return nullptr;
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}
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}
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}
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} // namespace webrtc
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