From a092cbf9b78f6518378cb72c936f3c344cc18818 Mon Sep 17 00:00:00 2001 From: "tina.legrand@webrtc.org" Date: Thu, 14 Feb 2013 09:28:10 +0000 Subject: [PATCH] Fixing lint warnings from previous commit In this CL I have removed (almost) all lint warnings I got for this commit: https://code.google.com/p/webrtc/source/detail?r=3454. The only warning not fixed is a warning about usage of non-const reference. This will be fixed in a separate CL. BUG=issue1372 Review URL: https://webrtc-codereview.appspot.com/1091006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3510 4adac7df-926f-26a2-2b94-8c16560cd09d --- .../main/source/acm_codec_database.cc | 63 +++----------- .../main/source/acm_codec_database.h | 6 -- .../audio_coding/main/source/acm_opus.cc | 2 +- .../main/source/audio_coding_module.cc | 6 +- .../main/source/audio_coding_module_impl.cc | 13 ++- .../audio_coding/main/test/TestAllCodecs.cc | 19 ++--- .../audio_coding/main/test/TestStereo.cc | 82 ++++++++++--------- 7 files changed, 72 insertions(+), 119 deletions(-) diff --git a/webrtc/modules/audio_coding/main/source/acm_codec_database.cc b/webrtc/modules/audio_coding/main/source/acm_codec_database.cc index fa7688f93..6a650c7eb 100644 --- a/webrtc/modules/audio_coding/main/source/acm_codec_database.cc +++ b/webrtc/modules/audio_coding/main/source/acm_codec_database.cc @@ -17,8 +17,6 @@ // references, where appropriate. #include "webrtc/modules/audio_coding/main/source/acm_codec_database.h" -#include - #include "webrtc/modules/audio_coding/main/source/acm_common_defs.h" #include "webrtc/system_wrappers/interface/trace.h" @@ -34,11 +32,11 @@ #include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h" #ifdef WEBRTC_CODEC_ISAC #include "webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h" -#include "webrtc/modules/audio_coding/main/source/acm_isac.h" -#include "webrtc/modules/audio_coding/main/source/acm_isac_macros.h" #endif #ifdef WEBRTC_CODEC_ISACFX #include "webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h" +#endif +#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) #include "webrtc/modules/audio_coding/main/source/acm_isac.h" #include "webrtc/modules/audio_coding/main/source/acm_isac_macros.h" #endif @@ -51,15 +49,15 @@ #include "webrtc/modules/audio_coding/main/source/acm_ilbc.h" #endif #ifdef WEBRTC_CODEC_AMR -#include "amr_interface.h" +#include "webrtc/modules/audio_coding/codecs/amr/include/amr_interface.h" #include "webrtc/modules/audio_coding/main/source/acm_amr.h" #endif #ifdef WEBRTC_CODEC_AMRWB -#include "amrwb_interface.h" +#include "webrtc/modules/audio_coding/codecs/amrwb/include/amrwb_interface.h" #include "webrtc/modules/audio_coding/main/source/acm_amrwb.h" #endif #ifdef WEBRTC_CODEC_CELT -#include "celt_interface.h" +#include "webrtc/modules/audio_coding/codecs/celt/include/celt_interface.h" #include "webrtc/modules/audio_coding/main/source/acm_celt.h" #endif #ifdef WEBRTC_CODEC_G722 @@ -67,23 +65,23 @@ #include "webrtc/modules/audio_coding/main/source/acm_g722.h" #endif #ifdef WEBRTC_CODEC_G722_1 -#include "g7221_interface.h" +#include "webrtc/modules/audio_coding/codecs/g7221/include/g7221_interface.h" #include "webrtc/modules/audio_coding/main/source/acm_g7221.h" #endif #ifdef WEBRTC_CODEC_G722_1C -#include "g7221c_interface.h" +#include "webrtc/modules/audio_coding/codecs/g7221c/include/g7221c_interface.h" #include "webrtc/modules/audio_coding/main/source/acm_g7221c.h" #endif #ifdef WEBRTC_CODEC_G729 -#include "g729_interface.h" +#include "webrtc/modules/audio_coding/codecs/g729/include/g729_interface.h" #include "webrtc/modules/audio_coding/main/source/acm_g729.h" #endif #ifdef WEBRTC_CODEC_G729_1 -#include "g7291_interface.h" +#include "webrtc/modules/audio_coding/codecs/g7291/include/g7291_interface.h" #include "webrtc/modules/audio_coding/main/source/acm_g7291.h" #endif #ifdef WEBRTC_CODEC_GSMFR -#include "gsmfr_interface.h" +#include "webrtc/modules/audio_coding/codecs/gsmfr/include/gsmfr_interface.h" #include "webrtc/modules/audio_coding/main/source/acm_gsmfr.h" #endif #ifdef WEBRTC_CODEC_OPUS @@ -91,7 +89,7 @@ #include "webrtc/modules/audio_coding/main/source/acm_opus.h" #endif #ifdef WEBRTC_CODEC_SPEEX -#include "speex_interface.h" +#include "webrtc/modules/audio_coding/codecs/speex/include/speex_interface.h" #include "webrtc/modules/audio_coding/main/source/acm_speex.h" #endif #ifdef WEBRTC_CODEC_AVT @@ -417,45 +415,6 @@ enum { kInvalidRate = -50 }; -// Gets the codec id number from the database. If there is some mismatch in -// the codec settings, an error message will be recorded in the error string. -// NOTE! Only the first mismatch found will be recorded in the error string. -int ACMCodecDB::CodecNumber(const CodecInst* codec_inst, int* mirror_id, - char* err_message, int max_message_len_byte) { - int codec_id = ACMCodecDB::CodecNumber(codec_inst, mirror_id); - - // Write error message if ACMCodecDB::CodecNumber() returned error. - if ((codec_id < 0) && (err_message != NULL)) { - char my_err_msg[1000]; - - if (codec_id == kInvalidCodec) { - sprintf(my_err_msg, "Call to ACMCodecDB::CodecNumber failed, Codec not " - "found"); - } else if (codec_id == kInvalidPayloadtype) { - sprintf(my_err_msg, "Call to ACMCodecDB::CodecNumber failed, payload " - "number %d is out of range for %s", codec_inst->pltype, - codec_inst->plname); - } else if (codec_id == kInvalidPacketSize) { - sprintf(my_err_msg, "Call to ACMCodecDB::CodecNumber failed, Packet " - "size is out of range for %s", codec_inst->plname); - } else if (codec_id == kInvalidRate) { - sprintf(my_err_msg, "Call to ACMCodecDB::CodecNumber failed, rate=%d " - "is not a valid rate for %s", codec_inst->rate, - codec_inst->plname); - } else { - // Other error - sprintf(my_err_msg, "invalid codec parameters to be registered, " - "ACMCodecDB::CodecNumber failed"); - } - - strncpy(err_message, my_err_msg, max_message_len_byte - 1); - // make sure that the message is null-terminated. - err_message[max_message_len_byte - 1] = '\0'; - } - - return codec_id; -} - // Gets the codec id number from the database. If there is some mismatch in // the codec settings, the function will return an error code. // NOTE! The first mismatch found will generate the return value. diff --git a/webrtc/modules/audio_coding/main/source/acm_codec_database.h b/webrtc/modules/audio_coding/main/source/acm_codec_database.h index 0ea7741c5..55f08d177 100644 --- a/webrtc/modules/audio_coding/main/source/acm_codec_database.h +++ b/webrtc/modules/audio_coding/main/source/acm_codec_database.h @@ -240,14 +240,8 @@ class ACMCodecDB { // Output: // [mirror_id] - mirror id, which most often is the same as the return // value, see above. - // [err_message] - if present, in the event of a mismatch found between the - // input and the database, a descriptive error message is - // written here. - // [err_message] - if present, the length of error message is returned here. // Return: // codec id if successful, otherwise < 0. - static int CodecNumber(const CodecInst* codec_inst, int* mirror_id, - char* err_message, int max_message_len_byte); static int CodecNumber(const CodecInst* codec_inst, int* mirror_id); static int CodecId(const CodecInst* codec_inst); static int CodecId(const char* payload_name, int frequency, int channels); diff --git a/webrtc/modules/audio_coding/main/source/acm_opus.cc b/webrtc/modules/audio_coding/main/source/acm_opus.cc index 5648ee329..8ea5d51d5 100644 --- a/webrtc/modules/audio_coding/main/source/acm_opus.cc +++ b/webrtc/modules/audio_coding/main/source/acm_opus.cc @@ -270,7 +270,7 @@ void ACMOpus::DestructDecoderSafe() { void ACMOpus::InternalDestructEncoderInst(void* ptr_inst) { if (ptr_inst != NULL) { - WebRtcOpus_EncoderFree((OpusEncInst*) ptr_inst); + WebRtcOpus_EncoderFree(reinterpret_cast(ptr_inst)); } return; } diff --git a/webrtc/modules/audio_coding/main/source/audio_coding_module.cc b/webrtc/modules/audio_coding/main/source/audio_coding_module.cc index 68913059b..dc69762c2 100644 --- a/webrtc/modules/audio_coding/main/source/audio_coding_module.cc +++ b/webrtc/modules/audio_coding/main/source/audio_coding_module.cc @@ -78,12 +78,12 @@ WebRtc_Word32 AudioCodingModule::Codec(const char* payload_name, // Checks the validity of the parameters of the given codec bool AudioCodingModule::IsCodecValid(const CodecInst& codec) { int mirror_id; - char err_msg[500]; - int codec_number = ACMCodecDB::CodecNumber(&codec, &mirror_id, err_msg, 500); + int codec_number = ACMCodecDB::CodecNumber(&codec, &mirror_id); if (codec_number < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, -1, err_msg); + WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, -1, + "Invalid codec settings."); return false; } else { return true; diff --git a/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc index 99761c027..4211be8b4 100644 --- a/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc +++ b/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc @@ -274,7 +274,7 @@ WebRtc_Word32 AudioCodingModuleImpl::ChangeUniqueId(const WebRtc_Word32 id) { CriticalSectionScoped lock(acm_crit_sect_); id_ = id; - for (int i = 0; i < ACMCodecDB::kMaxNumCodecs; i++) { + for (int i = 0; i < ACMCodecDB::kMaxNumCodecs; i++) { if (codecs_[i] != NULL) { codecs_[i]->SetUniqueID(id); } @@ -802,12 +802,10 @@ static int IsValidSendCodec(const CodecInst& send_codec, return -1; } - char error_message[500]; - int codec_id = ACMCodecDB::CodecNumber(&send_codec, mirror_id, error_message, - sizeof(error_message)); + int codec_id = ACMCodecDB::CodecNumber(&send_codec, mirror_id); if (codec_id < 0) { WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, acm_id, - error_message); + "Invalid settings for the send codec."); return -1; } @@ -1471,7 +1469,8 @@ int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame, timestamp_diff = in_frame.timestamp_ - last_in_timestamp_; } preprocess_frame_.timestamp_ = last_timestamp_ + - (WebRtc_UWord32)(timestamp_diff * ((double) send_codec_inst_.plfreq / + static_cast(timestamp_diff * + (static_cast(send_codec_inst_.plfreq) / static_cast(in_frame.sample_rate_hz_))); preprocess_frame_.samples_per_channel_ = input_resampler_.Resample10Msec( @@ -1546,7 +1545,7 @@ int AudioCodingModuleImpl::SetVADSafe(bool enable_dtx, && (mode != VADAggr) && (mode != VADVeryAggr)) { WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, "Invalid VAD Mode %d, no change is made to VAD/DTX status", - (int) mode); + static_cast(mode)); return -1; } diff --git a/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc b/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc index 5a590535a..46a589779 100644 --- a/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc +++ b/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc @@ -8,21 +8,21 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "TestAllCodecs.h" +#include "webrtc/modules/audio_coding/main/test/TestAllCodecs.h" #include #include #include "gtest/gtest.h" -#include "audio_coding_module.h" -#include "audio_coding_module_typedefs.h" -#include "common_types.h" -#include "engine_configurations.h" -#include "testsupport/fileutils.h" -#include "trace.h" -#include "typedefs.h" -#include "utility.h" +#include "webrtc/common_types.h" +#include "webrtc/engine_configurations.h" +#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h" +#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h" +#include "webrtc/modules/audio_coding/main/test/utility.h" +#include "webrtc/system_wrappers/interface/trace.h" +#include "webrtc/test/testsupport/fileutils.h" +#include "webrtc/typedefs.h" // Description of the test: // In this test we set up a one-way communication channel from a participant @@ -127,7 +127,6 @@ TestAllCodecs::~TestAllCodecs() { } void TestAllCodecs::Perform() { - const std::string file_name = webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"); infile_a_.Open(file_name, 32000, "rb"); diff --git a/webrtc/modules/audio_coding/main/test/TestStereo.cc b/webrtc/modules/audio_coding/main/test/TestStereo.cc index 52508e2a2..b06f19d7a 100644 --- a/webrtc/modules/audio_coding/main/test/TestStereo.cc +++ b/webrtc/modules/audio_coding/main/test/TestStereo.cc @@ -8,19 +8,19 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "TestStereo.h" +#include "webrtc/modules/audio_coding/main/test/TestStereo.h" #include -#include +#include #include "gtest/gtest.h" -#include "audio_coding_module_typedefs.h" -#include "common_types.h" -#include "engine_configurations.h" -#include "testsupport/fileutils.h" -#include "trace.h" -#include "utility.h" +#include "webrtc/common_types.h" +#include "webrtc/engine_configurations.h" +#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h" +#include "webrtc/modules/audio_coding/main/test/utility.h" +#include "webrtc/system_wrappers/interface/trace.h" +#include "webrtc/test/testsupport/fileutils.h" namespace webrtc { @@ -65,10 +65,10 @@ WebRtc_Word32 TestPackStereo::SendData( if (lost_packet_ == false) { if (frame_type != kAudioFrameCN) { rtp_info.type.Audio.isCNG = false; - rtp_info.type.Audio.channel = (int) codec_mode_; + rtp_info.type.Audio.channel = static_cast(codec_mode_); } else { rtp_info.type.Audio.isCNG = true; - rtp_info.type.Audio.channel = (int) kMono; + rtp_info.type.Audio.channel = static_cast(kMono); } status = receiver_acm_->IncomingPacket(payload_data, payload_size, rtp_info); @@ -245,7 +245,7 @@ void TestStereo::Perform() { #ifdef WEBRTC_CODEC_G722 if (test_mode_ != 0) { printf("===========================================================\n"); - printf("Test number: %d\n",test_cntr_ + 1); + printf("Test number: %d\n", test_cntr_ + 1); printf("Test type: Stereo-to-stereo\n"); } channel_a2b_->set_codec_mode(kStereo); @@ -275,7 +275,7 @@ void TestStereo::Perform() { #ifdef WEBRTC_CODEC_PCM16 if (test_mode_ != 0) { printf("===========================================================\n"); - printf("Test number: %d\n",test_cntr_ + 1); + printf("Test number: %d\n", test_cntr_ + 1); printf("Test type: Stereo-to-stereo\n"); } channel_a2b_->set_codec_mode(kStereo); @@ -298,7 +298,7 @@ void TestStereo::Perform() { if (test_mode_ != 0) { printf("===========================================================\n"); - printf("Test number: %d\n",test_cntr_ + 1); + printf("Test number: %d\n", test_cntr_ + 1); printf("Test type: Stereo-to-stereo\n"); } test_cntr_++; @@ -319,7 +319,7 @@ void TestStereo::Perform() { if (test_mode_ != 0) { printf("===========================================================\n"); - printf("Test number: %d\n",test_cntr_ + 1); + printf("Test number: %d\n", test_cntr_ + 1); printf("Test type: Stereo-to-stereo\n"); } test_cntr_++; @@ -414,7 +414,7 @@ void TestStereo::Perform() { #ifdef WEBRTC_CODEC_CELT if (test_mode_ != 0) { printf("===========================================================\n"); - printf("Test number: %d\n",test_cntr_ + 1); + printf("Test number: %d\n", test_cntr_ + 1); printf("Test type: Stereo-to-stereo\n"); } channel_a2b_->set_codec_mode(kStereo); @@ -437,7 +437,7 @@ void TestStereo::Perform() { #ifdef WEBRTC_CODEC_OPUS if (test_mode_ != 0) { printf("===========================================================\n"); - printf("Test number: %d\n",test_cntr_ + 1); + printf("Test number: %d\n", test_cntr_ + 1); printf("Test type: Stereo-to-stereo\n"); } channel_a2b_->set_codec_mode(kStereo); @@ -481,7 +481,7 @@ void TestStereo::Perform() { #ifdef WEBRTC_CODEC_G722 if (test_mode_ != 0) { printf("===============================================================\n"); - printf("Test number: %d\n",test_cntr_ + 1); + printf("Test number: %d\n", test_cntr_ + 1); printf("Test type: Mono-to-stereo\n"); } test_cntr_++; @@ -495,7 +495,7 @@ void TestStereo::Perform() { #ifdef WEBRTC_CODEC_PCM16 if (test_mode_ != 0) { printf("===============================================================\n"); - printf("Test number: %d\n",test_cntr_ + 1); + printf("Test number: %d\n", test_cntr_ + 1); printf("Test type: Mono-to-stereo\n"); } test_cntr_++; @@ -507,7 +507,7 @@ void TestStereo::Perform() { out_file_.Close(); if (test_mode_ != 0) { printf("===============================================================\n"); - printf("Test number: %d\n",test_cntr_ + 1); + printf("Test number: %d\n", test_cntr_ + 1); printf("Test type: Mono-to-stereo\n"); } test_cntr_++; @@ -518,7 +518,7 @@ void TestStereo::Perform() { out_file_.Close(); if (test_mode_ != 0) { printf("===============================================================\n"); - printf("Test number: %d\n",test_cntr_ + 1); + printf("Test number: %d\n", test_cntr_ + 1); printf("Test type: Mono-to-stereo\n"); } test_cntr_++; @@ -531,7 +531,7 @@ void TestStereo::Perform() { #ifdef PCMA_AND_PCMU if (test_mode_ != 0) { printf("===============================================================\n"); - printf("Test number: %d\n",test_cntr_ + 1); + printf("Test number: %d\n", test_cntr_ + 1); printf("Test type: Mono-to-stereo\n"); } test_cntr_++; @@ -548,7 +548,7 @@ void TestStereo::Perform() { #ifdef WEBRTC_CODEC_CELT if (test_mode_ != 0) { printf("===============================================================\n"); - printf("Test number: %d\n",test_cntr_ + 1); + printf("Test number: %d\n", test_cntr_ + 1); printf("Test type: Mono-to-stereo\n"); } test_cntr_++; @@ -562,7 +562,7 @@ void TestStereo::Perform() { #ifdef WEBRTC_CODEC_OPUS if (test_mode_ != 0) { printf("===============================================================\n"); - printf("Test number: %d\n",test_cntr_ + 1); + printf("Test number: %d\n", test_cntr_ + 1); printf("Test type: Mono-to-stereo\n"); } @@ -591,7 +591,7 @@ void TestStereo::Perform() { // Run stereo audio and mono codec. if (test_mode_ != 0) { printf("===============================================================\n"); - printf("Test number: %d\n",test_cntr_ + 1); + printf("Test number: %d\n", test_cntr_ + 1); printf("Test type: Stereo-to-mono\n"); } test_cntr_++; @@ -613,7 +613,7 @@ void TestStereo::Perform() { #ifdef WEBRTC_CODEC_PCM16 if (test_mode_ != 0) { printf("===============================================================\n"); - printf("Test number: %d\n",test_cntr_ + 1); + printf("Test number: %d\n", test_cntr_ + 1); printf("Test type: Stereo-to-mono\n"); } test_cntr_++; @@ -624,9 +624,9 @@ void TestStereo::Perform() { out_file_.Close(); if (test_mode_ != 0) { printf("===============================================================\n"); - printf("Test number: %d\n",test_cntr_ + 1); + printf("Test number: %d\n", test_cntr_ + 1); printf("Test type: Stereo-to-mono\n"); - } + } test_cntr_++; OpenOutFile(test_cntr_); RegisterSendCodec('A', codec_l16, 16000, 256000, 160, codec_channels, @@ -635,20 +635,20 @@ void TestStereo::Perform() { out_file_.Close(); if (test_mode_ != 0) { printf("==============================================================\n"); - printf("Test number: %d\n",test_cntr_ + 1); + printf("Test number: %d\n", test_cntr_ + 1); printf("Test type: Stereo-to-mono\n"); - } - test_cntr_++; - OpenOutFile(test_cntr_); - RegisterSendCodec('A', codec_l16, 32000, 512000, 320, codec_channels, - l16_32khz_pltype_); - Run(channel_a2b_, audio_channels, codec_channels); - out_file_.Close(); + } + test_cntr_++; + OpenOutFile(test_cntr_); + RegisterSendCodec('A', codec_l16, 32000, 512000, 320, codec_channels, + l16_32khz_pltype_); + Run(channel_a2b_, audio_channels, codec_channels); + out_file_.Close(); #endif #ifdef PCMA_AND_PCMU if (test_mode_ != 0) { printf("===============================================================\n"); - printf("Test number: %d\n",test_cntr_ + 1); + printf("Test number: %d\n", test_cntr_ + 1); printf("Test type: Stereo-to-mono\n"); } test_cntr_++; @@ -664,7 +664,7 @@ void TestStereo::Perform() { #ifdef WEBRTC_CODEC_CELT if (test_mode_ != 0) { printf("===============================================================\n"); - printf("Test number: %d\n",test_cntr_ + 1); + printf("Test number: %d\n", test_cntr_ + 1); printf("Test type: Stereo-to-mono\n"); } test_cntr_++; @@ -677,7 +677,7 @@ void TestStereo::Perform() { #ifdef WEBRTC_CODEC_OPUS if (test_mode_ != 0) { printf("===============================================================\n"); - printf("Test number: %d\n",test_cntr_ + 1); + printf("Test number: %d\n", test_cntr_ + 1); printf("Test type: Stereo-to-mono\n"); } test_cntr_++; @@ -794,11 +794,13 @@ void TestStereo::RegisterSendCodec(char side, char* codec_name, // For Celt the packet size in bytes is already counting the stereo part. if (!strcmp(codec_name, "CELT")) { pack_size_bytes_ = (WebRtc_UWord16)( - (float) (pack_size * rate) / (float) (sampling_freq_hz * 8) + 0.875) + static_cast(pack_size * rate) / + static_cast(sampling_freq_hz * 8) + 0.875) / channels; } else { pack_size_bytes_ = (WebRtc_UWord16)( - (float) (pack_size * rate) / (float) (sampling_freq_hz * 8) + 0.875); + static_cast(pack_size * rate) / + static_cast(sampling_freq_hz * 8) + 0.875); } // Set pointer to the ACM where to register the codec