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7120742701
Now that we have moved WebRTC from src/webrtc to src/, common_types.h and typedefs.h are triggering a cpplint error. The cpplint complaint is: Include the directory when naming .h files [build/include] [4] This CL disables the error but we have to remove these two headers from the root directory. NOPRESUBMIT=true Bug: webrtc:5876 Change-Id: I08e1b69aadcc4b28ab83bf25e3819d135d41d333 Reviewed-on: https://webrtc-review.googlesource.com/1577 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19859}
61 lines
2 KiB
C++
61 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/isac/audio_encoder_isac_fix.h"
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#include "common_types.h" // NOLINT(build/include)
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#include "modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h"
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#include "rtc_base/ptr_util.h"
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#include "rtc_base/string_to_number.h"
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namespace webrtc {
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rtc::Optional<AudioEncoderIsacFix::Config> AudioEncoderIsacFix::SdpToConfig(
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const SdpAudioFormat& format) {
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if (STR_CASE_CMP(format.name.c_str(), "ISAC") == 0 &&
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format.clockrate_hz == 16000 && format.num_channels == 1) {
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Config config;
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const 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 >= 60) {
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config.frame_size_ms = 60;
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}
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}
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return rtc::Optional<Config>(config);
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} else {
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return rtc::Optional<Config>();
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}
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}
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void AudioEncoderIsacFix::AppendSupportedEncoders(
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std::vector<AudioCodecSpec>* specs) {
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const SdpAudioFormat fmt = {"ISAC", 16000, 1};
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const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt));
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specs->push_back({fmt, info});
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}
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AudioCodecInfo AudioEncoderIsacFix::QueryAudioEncoder(
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AudioEncoderIsacFix::Config config) {
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RTC_DCHECK(config.IsOk());
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return {16000, 1, 32000, 10000, 32000};
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}
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std::unique_ptr<AudioEncoder> AudioEncoderIsacFix::MakeAudioEncoder(
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AudioEncoderIsacFix::Config config,
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int payload_type) {
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RTC_DCHECK(config.IsOk());
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AudioEncoderIsacFixImpl::Config c;
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c.frame_size_ms = config.frame_size_ms;
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c.payload_type = payload_type;
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return rtc::MakeUnique<AudioEncoderIsacFixImpl>(c);
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}
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} // namespace webrtc
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