6955870806
IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan. BUG=chromium:81439 TEST=none R=henrik.lundin@webrtc.org, henrika@webrtc.org, kjellander@webrtc.org, minyue@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org Review URL: https://codereview.webrtc.org/1316523002 . Cr-Commit-Position: refs/heads/master@{#11229}
99 lines
3.0 KiB
C++
99 lines
3.0 KiB
C++
/*
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* Copyright (c) 2012 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/modules/audio_coding/include/audio_coding_module.h"
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#include "webrtc/base/checks.h"
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#include "webrtc/common_types.h"
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#include "webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h"
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#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
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#include "webrtc/system_wrappers/include/clock.h"
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#include "webrtc/system_wrappers/include/trace.h"
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namespace webrtc {
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// Create module
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AudioCodingModule* AudioCodingModule::Create(int id) {
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Config config;
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config.id = id;
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config.clock = Clock::GetRealTimeClock();
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return Create(config);
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}
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AudioCodingModule* AudioCodingModule::Create(int id, Clock* clock) {
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Config config;
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config.id = id;
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config.clock = clock;
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return Create(config);
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}
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AudioCodingModule* AudioCodingModule::Create(const Config& config) {
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return new acm2::AudioCodingModuleImpl(config);
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}
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int AudioCodingModule::NumberOfCodecs() {
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return static_cast<int>(acm2::RentACodec::NumberOfCodecs());
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}
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int AudioCodingModule::Codec(int list_id, CodecInst* codec) {
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auto codec_id = acm2::RentACodec::CodecIdFromIndex(list_id);
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if (!codec_id)
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return -1;
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auto ci = acm2::RentACodec::CodecInstById(*codec_id);
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if (!ci)
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return -1;
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*codec = *ci;
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return 0;
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}
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int AudioCodingModule::Codec(const char* payload_name,
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CodecInst* codec,
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int sampling_freq_hz,
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size_t channels) {
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rtc::Optional<CodecInst> ci = acm2::RentACodec::CodecInstByParams(
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payload_name, sampling_freq_hz, channels);
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if (ci) {
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*codec = *ci;
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return 0;
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} else {
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// We couldn't find a matching codec, so set the parameters to unacceptable
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// values and return.
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codec->plname[0] = '\0';
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codec->pltype = -1;
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codec->pacsize = 0;
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codec->rate = 0;
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codec->plfreq = 0;
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return -1;
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}
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}
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int AudioCodingModule::Codec(const char* payload_name,
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int sampling_freq_hz,
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size_t channels) {
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rtc::Optional<acm2::RentACodec::CodecId> ci =
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acm2::RentACodec::CodecIdByParams(payload_name, sampling_freq_hz,
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channels);
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if (!ci)
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return -1;
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rtc::Optional<int> i = acm2::RentACodec::CodecIndexFromId(*ci);
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return i ? *i : -1;
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}
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// Checks the validity of the parameters of the given codec
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bool AudioCodingModule::IsCodecValid(const CodecInst& codec) {
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bool valid = acm2::RentACodec::IsCodecValid(codec);
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if (!valid)
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, -1,
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"Invalid codec setting");
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return valid;
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}
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} // namespace webrtc
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