webrtc_m130/webrtc/api/audio_codecs/g722/audio_encoder_g722.cc
Edward Lemur c20978e581 Rename webrtc/base -> webrtc/rtc_base
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL.
NOTRY=True
NOTREECHECKS=True
TBR=kwiberg@webrtc.org, kjellander@webrtc.org

Bug: webrtc:7634
Change-Id: I3cca0fbaa807b563c95979cccd6d1bec32055f36
Reviewed-on: https://chromium-review.googlesource.com/562156
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18919}
2017-07-06 19:11:40 +00:00

49 lines
1.5 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 "webrtc/api/audio_codecs/g722/audio_encoder_g722.h"
#include <memory>
#include <vector>
#include "webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h"
#include "webrtc/rtc_base/ptr_util.h"
#include "webrtc/rtc_base/safe_conversions.h"
namespace webrtc {
rtc::Optional<AudioEncoderG722Config> AudioEncoderG722::SdpToConfig(
const SdpAudioFormat& format) {
return AudioEncoderG722Impl::SdpToConfig(format);
}
void AudioEncoderG722::AppendSupportedEncoders(
std::vector<AudioCodecSpec>* specs) {
const SdpAudioFormat fmt = {"g722", 8000, 1};
const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt));
specs->push_back({fmt, info});
}
AudioCodecInfo AudioEncoderG722::QueryAudioEncoder(
const AudioEncoderG722Config& config) {
RTC_DCHECK(config.IsOk());
return {16000, rtc::dchecked_cast<size_t>(config.num_channels),
64000 * config.num_channels};
}
std::unique_ptr<AudioEncoder> AudioEncoderG722::MakeAudioEncoder(
const AudioEncoderG722Config& config,
int payload_type) {
RTC_DCHECK(config.IsOk());
return rtc::MakeUnique<AudioEncoderG722Impl>(config, payload_type);
}
} // namespace webrtc