Reformat the WebRTC code base
Running clang-format with chromium's style guide. The goal is n-fold: * providing consistency and readability (that's what code guidelines are for) * preventing noise with presubmit checks and git cl format * building on the previous point: making it easier to automatically fix format issues * you name it Please consider using git-hyper-blame to ignore this commit. Bug: webrtc:9340 Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87 Reviewed-on: https://webrtc-review.googlesource.com/81185 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23660}
This commit is contained in:
@ -42,7 +42,7 @@ bool IsILBCRateValid(int rate, int frame_size_samples) {
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(rate == 13300)) {
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return true;
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} else if (((frame_size_samples == 160) || (frame_size_samples == 320)) &&
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(rate == 15200)) {
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(rate == 15200)) {
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return true;
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} else {
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return false;
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@ -62,55 +62,54 @@ bool IsOpusRateValid(int rate) {
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const CodecInst ACMCodecDB::database_[] = {
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#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
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{103, "ISAC", 16000, 480, 1, 32000},
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# if (defined(WEBRTC_CODEC_ISAC))
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{104, "ISAC", 32000, 960, 1, 56000},
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# endif
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{103, "ISAC", 16000, 480, 1, 32000},
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#if (defined(WEBRTC_CODEC_ISAC))
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{104, "ISAC", 32000, 960, 1, 56000},
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#endif
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// Mono
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{107, "L16", 8000, 80, 1, 128000},
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{108, "L16", 16000, 160, 1, 256000},
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{109, "L16", 32000, 320, 1, 512000},
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// Stereo
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{111, "L16", 8000, 80, 2, 128000},
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{112, "L16", 16000, 160, 2, 256000},
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{113, "L16", 32000, 320, 2, 512000},
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// G.711, PCM mu-law and A-law.
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// Mono
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{0, "PCMU", 8000, 160, 1, 64000},
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{8, "PCMA", 8000, 160, 1, 64000},
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// Stereo
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{110, "PCMU", 8000, 160, 2, 64000},
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{118, "PCMA", 8000, 160, 2, 64000},
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#endif
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// Mono
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{107, "L16", 8000, 80, 1, 128000},
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{108, "L16", 16000, 160, 1, 256000},
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{109, "L16", 32000, 320, 1, 512000},
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// Stereo
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{111, "L16", 8000, 80, 2, 128000},
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{112, "L16", 16000, 160, 2, 256000},
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{113, "L16", 32000, 320, 2, 512000},
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// G.711, PCM mu-law and A-law.
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// Mono
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{0, "PCMU", 8000, 160, 1, 64000},
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{8, "PCMA", 8000, 160, 1, 64000},
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// Stereo
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{110, "PCMU", 8000, 160, 2, 64000},
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{118, "PCMA", 8000, 160, 2, 64000},
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#ifdef WEBRTC_CODEC_ILBC
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{102, "ILBC", 8000, 240, 1, 13300},
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{102, "ILBC", 8000, 240, 1, 13300},
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#endif
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// Mono
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{9, "G722", 16000, 320, 1, 64000},
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// Stereo
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{119, "G722", 16000, 320, 2, 64000},
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// Mono
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{9, "G722", 16000, 320, 1, 64000},
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// Stereo
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{119, "G722", 16000, 320, 2, 64000},
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#ifdef WEBRTC_CODEC_OPUS
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// Opus internally supports 48, 24, 16, 12, 8 kHz.
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// Mono and stereo.
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{120, "opus", 48000, 960, 2, 64000},
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// Opus internally supports 48, 24, 16, 12, 8 kHz.
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// Mono and stereo.
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{120, "opus", 48000, 960, 2, 64000},
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#endif
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// Comfort noise for four different sampling frequencies.
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{13, "CN", 8000, 240, 1, 0},
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{98, "CN", 16000, 480, 1, 0},
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{99, "CN", 32000, 960, 1, 0},
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// Comfort noise for four different sampling frequencies.
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{13, "CN", 8000, 240, 1, 0},
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{98, "CN", 16000, 480, 1, 0},
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{99, "CN", 32000, 960, 1, 0},
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#ifdef ENABLE_48000_HZ
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{100, "CN", 48000, 1440, 1, 0},
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{100, "CN", 48000, 1440, 1, 0},
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#endif
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{106, "telephone-event", 8000, 240, 1, 0},
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{114, "telephone-event", 16000, 240, 1, 0},
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{115, "telephone-event", 32000, 240, 1, 0},
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{116, "telephone-event", 48000, 240, 1, 0},
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{106, "telephone-event", 8000, 240, 1, 0},
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{114, "telephone-event", 16000, 240, 1, 0},
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{115, "telephone-event", 32000, 240, 1, 0},
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{116, "telephone-event", 48000, 240, 1, 0},
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#ifdef WEBRTC_CODEC_RED
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{127, "red", 8000, 0, 1, 0},
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{127, "red", 8000, 0, 1, 0},
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#endif
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// To prevent compile errors due to trailing commas.
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{-1, "Null", -1, -1, 0, -1}
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};
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// To prevent compile errors due to trailing commas.
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{-1, "Null", -1, -1, 0, -1}};
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// Create database with all codec settings at compile time.
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// Each entry needs the following parameters in the given order:
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@ -119,9 +118,9 @@ const CodecInst ACMCodecDB::database_[] = {
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const ACMCodecDB::CodecSettings ACMCodecDB::codec_settings_[] = {
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#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
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{2, {480, 960}, 0, 1},
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# if (defined(WEBRTC_CODEC_ISAC))
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#if (defined(WEBRTC_CODEC_ISAC))
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{1, {960}, 0, 1},
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# endif
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#endif
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#endif
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// Mono
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{4, {80, 160, 240, 320}, 0, 2},
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@ -146,9 +145,9 @@ const ACMCodecDB::CodecSettings ACMCodecDB::codec_settings_[] = {
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// Stereo
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{6, {160, 320, 480, 640, 800, 960}, 0, 2},
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#ifdef WEBRTC_CODEC_OPUS
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// Opus supports frames shorter than 10ms,
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// but it doesn't help us to use them.
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// Mono and stereo.
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// Opus supports frames shorter than 10ms,
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// but it doesn't help us to use them.
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// Mono and stereo.
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#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
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{5, {480, 960, 1920, 2880, 5760}, 0, 2},
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#else
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@ -171,16 +170,15 @@ const ACMCodecDB::CodecSettings ACMCodecDB::codec_settings_[] = {
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{1, {0}, 0, 1},
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#endif
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// To prevent compile errors due to trailing commas.
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{-1, {-1}, -1, 0}
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};
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{-1, {-1}, -1, 0}};
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// Create a database of all NetEQ decoders at compile time.
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const NetEqDecoder ACMCodecDB::neteq_decoders_[] = {
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#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
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NetEqDecoder::kDecoderISAC,
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# if (defined(WEBRTC_CODEC_ISAC))
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#if (defined(WEBRTC_CODEC_ISAC))
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NetEqDecoder::kDecoderISACswb,
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# endif
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#endif
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#endif
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// Mono
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NetEqDecoder::kDecoderPCM16B, NetEqDecoder::kDecoderPCM16Bwb,
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@ -210,10 +208,8 @@ const NetEqDecoder ACMCodecDB::neteq_decoders_[] = {
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#ifdef ENABLE_48000_HZ
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NetEqDecoder::kDecoderCNGswb48kHz,
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#endif
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NetEqDecoder::kDecoderAVT,
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NetEqDecoder::kDecoderAVT16kHz,
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NetEqDecoder::kDecoderAVT32kHz,
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NetEqDecoder::kDecoderAVT48kHz,
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NetEqDecoder::kDecoderAVT, NetEqDecoder::kDecoderAVT16kHz,
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NetEqDecoder::kDecoderAVT32kHz, NetEqDecoder::kDecoderAVT48kHz,
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#ifdef WEBRTC_CODEC_RED
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NetEqDecoder::kDecoderRED,
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#endif
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@ -260,8 +256,7 @@ int ACMCodecDB::CodecNumber(const CodecInst& codec_inst) {
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int i;
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int packet_size_samples;
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for (i = 0; i < codec_settings_[codec_id].num_packet_sizes; i++) {
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packet_size_samples =
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codec_settings_[codec_id].packet_sizes_samples[i];
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packet_size_samples = codec_settings_[codec_id].packet_sizes_samples[i];
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if (codec_inst.pacsize == packet_size_samples) {
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packet_size_ok = true;
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break;
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@ -282,11 +277,10 @@ int ACMCodecDB::CodecNumber(const CodecInst& codec_inst) {
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if (STR_CASE_CMP("isac", codec_inst.plname) == 0) {
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return IsISACRateValid(codec_inst.rate) ? codec_id : kInvalidRate;
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} else if (STR_CASE_CMP("ilbc", codec_inst.plname) == 0) {
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return IsILBCRateValid(codec_inst.rate, codec_inst.pacsize)
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? codec_id : kInvalidRate;
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return IsILBCRateValid(codec_inst.rate, codec_inst.pacsize) ? codec_id
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: kInvalidRate;
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} else if (STR_CASE_CMP("opus", codec_inst.plname) == 0) {
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return IsOpusRateValid(codec_inst.rate)
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? codec_id : kInvalidRate;
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return IsOpusRateValid(codec_inst.rate) ? codec_id : kInvalidRate;
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}
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return database_[codec_id].rate == codec_inst.rate ? codec_id : kInvalidRate;
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@ -298,8 +292,7 @@ int ACMCodecDB::CodecNumber(const CodecInst& codec_inst) {
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// Does not check other codec settings, such as payload type and packet size.
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// Returns the id of the codec, or -1 if no match is found.
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int ACMCodecDB::CodecId(const CodecInst& codec_inst) {
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return (CodecId(codec_inst.plname, codec_inst.plfreq,
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codec_inst.channels));
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return (CodecId(codec_inst.plname, codec_inst.plfreq, codec_inst.channels));
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}
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int ACMCodecDB::CodecId(const char* payload_name,
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