mirror of
https://github.com/mollyim/webrtc.git
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71 lines
2.5 KiB
C++
71 lines
2.5 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/L16/audio_encoder_L16.h"
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#include <memory>
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#include "absl/strings/match.h"
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#include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
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#include "modules/audio_coding/codecs/pcm16b/pcm16b_common.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<AudioEncoderL16::Config> AudioEncoderL16::SdpToConfig(
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const SdpAudioFormat& format) {
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if (!rtc::IsValueInRangeForNumericType<int>(format.num_channels)) {
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return absl::nullopt;
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}
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Config config;
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config.sample_rate_hz = format.clockrate_hz;
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config.num_channels = rtc::dchecked_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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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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return absl::EqualsIgnoreCase(format.name, "L16") && config.IsOk()
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? absl::optional<Config>(config)
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: absl::nullopt;
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}
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void AudioEncoderL16::AppendSupportedEncoders(
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std::vector<AudioCodecSpec>* specs) {
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// RingRTC change to unused audio codecs
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// Pcm16BAppendSupportedCodecSpecs(specs);
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}
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AudioCodecInfo AudioEncoderL16::QueryAudioEncoder(
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const AudioEncoderL16::Config& config) {
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RTC_DCHECK(config.IsOk());
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return {config.sample_rate_hz,
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rtc::dchecked_cast<size_t>(config.num_channels),
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config.sample_rate_hz * config.num_channels * 16};
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}
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std::unique_ptr<AudioEncoder> AudioEncoderL16::MakeAudioEncoder(
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const AudioEncoderL16::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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AudioEncoderPcm16B::Config c;
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c.sample_rate_hz = config.sample_rate_hz;
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c.num_channels = config.num_channels;
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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 std::make_unique<AudioEncoderPcm16B>(c);
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}
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} // namespace webrtc
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