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bb547203bf
In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
67 lines
2.3 KiB
C++
67 lines
2.3 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 "webrtc/api/audio_codecs/g722/audio_encoder_g722.h"
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#include <memory>
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#include <vector>
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#include "webrtc/common_types.h"
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#include "webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h"
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#include "webrtc/rtc_base/ptr_util.h"
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#include "webrtc/rtc_base/safe_conversions.h"
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#include "webrtc/rtc_base/safe_minmax.h"
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#include "webrtc/rtc_base/string_to_number.h"
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namespace webrtc {
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rtc::Optional<AudioEncoderG722Config> AudioEncoderG722::SdpToConfig(
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const SdpAudioFormat& format) {
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if (STR_CASE_CMP(format.name.c_str(), "g722") != 0 ||
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format.clockrate_hz != 8000) {
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return rtc::Optional<AudioEncoderG722Config>();
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}
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AudioEncoderG722Config config;
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config.num_channels = rtc::checked_cast<int>(format.num_channels);
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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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auto ptime = rtc::StringToNumber<int>(ptime_iter->second);
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if (ptime && *ptime > 0) {
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const int whole_packets = *ptime / 10;
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config.frame_size_ms = rtc::SafeClamp<int>(whole_packets * 10, 10, 60);
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}
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}
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return config.IsOk() ? rtc::Optional<AudioEncoderG722Config>(config)
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: rtc::Optional<AudioEncoderG722Config>();
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}
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void AudioEncoderG722::AppendSupportedEncoders(
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std::vector<AudioCodecSpec>* specs) {
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const SdpAudioFormat fmt = {"G722", 8000, 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 AudioEncoderG722::QueryAudioEncoder(
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const AudioEncoderG722Config& config) {
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RTC_DCHECK(config.IsOk());
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return {16000, 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> AudioEncoderG722::MakeAudioEncoder(
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const AudioEncoderG722Config& config,
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int payload_type) {
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RTC_DCHECK(config.IsOk());
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return rtc::MakeUnique<AudioEncoderG722Impl>(config, payload_type);
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}
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} // namespace webrtc
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