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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}
63 lines
2 KiB
C++
63 lines
2 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/opus/audio_decoder_opus.h"
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#include <memory>
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#include <utility>
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#include <vector>
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#include "common_types.h" // NOLINT(build/include)
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#include "modules/audio_coding/codecs/opus/audio_decoder_opus.h"
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#include "rtc_base/ptr_util.h"
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namespace webrtc {
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absl::optional<AudioDecoderOpus::Config> AudioDecoderOpus::SdpToConfig(
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const SdpAudioFormat& format) {
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const auto num_channels = [&]() -> absl::optional<int> {
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auto stereo = format.parameters.find("stereo");
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if (stereo != format.parameters.end()) {
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if (stereo->second == "0") {
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return 1;
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} else if (stereo->second == "1") {
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return 2;
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} else {
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return absl::nullopt; // Bad stereo parameter.
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}
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}
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return 1; // Default to mono.
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}();
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if (STR_CASE_CMP(format.name.c_str(), "opus") == 0 &&
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format.clockrate_hz == 48000 && format.num_channels == 2 &&
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num_channels) {
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return Config{*num_channels};
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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 AudioDecoderOpus::AppendSupportedDecoders(
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std::vector<AudioCodecSpec>* specs) {
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AudioCodecInfo opus_info{48000, 1, 64000, 6000, 510000};
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opus_info.allow_comfort_noise = false;
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opus_info.supports_network_adaption = true;
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SdpAudioFormat opus_format(
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{"opus", 48000, 2, {{"minptime", "10"}, {"useinbandfec", "1"}}});
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specs->push_back({std::move(opus_format), std::move(opus_info)});
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}
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std::unique_ptr<AudioDecoder> AudioDecoderOpus::MakeAudioDecoder(
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Config config,
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absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
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return rtc::MakeUnique<AudioDecoderOpusImpl>(config.num_channels);
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}
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} // namespace webrtc
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