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Unified Diff: webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc

Issue 1481493004: audio_coding: remove "main" directory (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebased Created 5 years, 1 month ago
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Index: webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc
deleted file mode 100644
index 889d62092a03965bf2c2601b1fa31c8191f53ab2..0000000000000000000000000000000000000000
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc
+++ /dev/null
@@ -1,98 +0,0 @@
-/*
- * Copyright (c) 2012 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/modules/audio_coding/main/include/audio_coding_module.h"
-
-#include "webrtc/base/checks.h"
-#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h"
-#include "webrtc/modules/audio_coding/main/acm2/rent_a_codec.h"
-#include "webrtc/system_wrappers/include/clock.h"
-#include "webrtc/system_wrappers/include/trace.h"
-
-namespace webrtc {
-
-// Create module
-AudioCodingModule* AudioCodingModule::Create(int id) {
- Config config;
- config.id = id;
- config.clock = Clock::GetRealTimeClock();
- return Create(config);
-}
-
-AudioCodingModule* AudioCodingModule::Create(int id, Clock* clock) {
- Config config;
- config.id = id;
- config.clock = clock;
- return Create(config);
-}
-
-AudioCodingModule* AudioCodingModule::Create(const Config& config) {
- return new acm2::AudioCodingModuleImpl(config);
-}
-
-int AudioCodingModule::NumberOfCodecs() {
- return static_cast<int>(acm2::RentACodec::NumberOfCodecs());
-}
-
-int AudioCodingModule::Codec(int list_id, CodecInst* codec) {
- auto codec_id = acm2::RentACodec::CodecIdFromIndex(list_id);
- if (!codec_id)
- return -1;
- auto ci = acm2::RentACodec::CodecInstById(*codec_id);
- if (!ci)
- return -1;
- *codec = *ci;
- return 0;
-}
-
-int AudioCodingModule::Codec(const char* payload_name,
- CodecInst* codec,
- int sampling_freq_hz,
- int channels) {
- rtc::Optional<CodecInst> ci = acm2::RentACodec::CodecInstByParams(
- payload_name, sampling_freq_hz, channels);
- if (ci) {
- *codec = *ci;
- return 0;
- } else {
- // We couldn't find a matching codec, so set the parameters to unacceptable
- // values and return.
- codec->plname[0] = '\0';
- codec->pltype = -1;
- codec->pacsize = 0;
- codec->rate = 0;
- codec->plfreq = 0;
- return -1;
- }
-}
-
-int AudioCodingModule::Codec(const char* payload_name,
- int sampling_freq_hz,
- int channels) {
- rtc::Optional<acm2::RentACodec::CodecId> ci =
- acm2::RentACodec::CodecIdByParams(payload_name, sampling_freq_hz,
- channels);
- if (!ci)
- return -1;
- rtc::Optional<int> i = acm2::RentACodec::CodecIndexFromId(*ci);
- return i ? *i : -1;
-}
-
-// Checks the validity of the parameters of the given codec
-bool AudioCodingModule::IsCodecValid(const CodecInst& codec) {
- bool valid = acm2::RentACodec::IsCodecValid(codec);
- if (!valid)
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, -1,
- "Invalid codec setting");
- return valid;
-}
-
-} // namespace webrtc

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