From d1d79f6866fcde94bb0354cec7d6ecaaf72de235 Mon Sep 17 00:00:00 2001 From: kwiberg Date: Fri, 25 Aug 2017 22:22:42 -0700 Subject: [PATCH] Remove dead code This code became dead when the builtin audio codec factories were rewritten in https://codereview.webrtc.org/2997713002/. BUG=webrtc:7821, webrtc:7822 Review-Url: https://codereview.webrtc.org/3003603002 Cr-Commit-Position: refs/heads/master@{#19535} --- webrtc/api/audio_codecs/g722/BUILD.gn | 1 + .../audio_codecs/g722/audio_encoder_g722.cc | 21 +++++++- webrtc/api/audio_codecs/ilbc/BUILD.gn | 1 + .../audio_codecs/ilbc/audio_encoder_ilbc.cc | 20 ++++++- .../codecs/g711/audio_encoder_pcm.cc | 48 ----------------- .../codecs/g711/audio_encoder_pcm.h | 11 ---- .../codecs/g722/audio_encoder_g722.cc | 40 -------------- .../codecs/g722/audio_encoder_g722.h | 9 ---- .../codecs/ilbc/audio_encoder_ilbc.cc | 40 -------------- .../codecs/ilbc/audio_encoder_ilbc.h | 9 ---- .../codecs/isac/audio_encoder_isac_t.h | 6 --- .../codecs/isac/audio_encoder_isac_t_impl.h | 52 ------------------- .../codecs/pcm16b/audio_encoder_pcm16b.cc | 39 +------------- .../codecs/pcm16b/audio_encoder_pcm16b.h | 5 -- 14 files changed, 42 insertions(+), 260 deletions(-) diff --git a/webrtc/api/audio_codecs/g722/BUILD.gn b/webrtc/api/audio_codecs/g722/BUILD.gn index 2c1349a7c5..f3108e7b21 100644 --- a/webrtc/api/audio_codecs/g722/BUILD.gn +++ b/webrtc/api/audio_codecs/g722/BUILD.gn @@ -26,6 +26,7 @@ rtc_static_library("audio_encoder_g722") { deps = [ ":audio_encoder_g722_config", "..:audio_codecs_api", + "../../..:webrtc_common", "../../../modules/audio_coding:g722", "../../../rtc_base:rtc_base_approved", ] diff --git a/webrtc/api/audio_codecs/g722/audio_encoder_g722.cc b/webrtc/api/audio_codecs/g722/audio_encoder_g722.cc index 09b3faf745..b9df585c5f 100644 --- a/webrtc/api/audio_codecs/g722/audio_encoder_g722.cc +++ b/webrtc/api/audio_codecs/g722/audio_encoder_g722.cc @@ -13,15 +13,34 @@ #include #include +#include "webrtc/common_types.h" #include "webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h" #include "webrtc/rtc_base/ptr_util.h" #include "webrtc/rtc_base/safe_conversions.h" +#include "webrtc/rtc_base/safe_minmax.h" +#include "webrtc/rtc_base/string_to_number.h" namespace webrtc { rtc::Optional AudioEncoderG722::SdpToConfig( const SdpAudioFormat& format) { - return AudioEncoderG722Impl::SdpToConfig(format); + if (STR_CASE_CMP(format.name.c_str(), "g722") != 0 || + format.clockrate_hz != 8000) { + return rtc::Optional(); + } + + AudioEncoderG722Config config; + config.num_channels = rtc::checked_cast(format.num_channels); + auto ptime_iter = format.parameters.find("ptime"); + if (ptime_iter != format.parameters.end()) { + auto ptime = rtc::StringToNumber(ptime_iter->second); + if (ptime && *ptime > 0) { + const int whole_packets = *ptime / 10; + config.frame_size_ms = rtc::SafeClamp(whole_packets * 10, 10, 60); + } + } + return config.IsOk() ? rtc::Optional(config) + : rtc::Optional(); } void AudioEncoderG722::AppendSupportedEncoders( diff --git a/webrtc/api/audio_codecs/ilbc/BUILD.gn b/webrtc/api/audio_codecs/ilbc/BUILD.gn index 6ef8856639..ab9681a3e5 100644 --- a/webrtc/api/audio_codecs/ilbc/BUILD.gn +++ b/webrtc/api/audio_codecs/ilbc/BUILD.gn @@ -26,6 +26,7 @@ rtc_static_library("audio_encoder_ilbc") { deps = [ ":audio_encoder_ilbc_config", "..:audio_codecs_api", + "../../..:webrtc_common", "../../../modules/audio_coding:ilbc", "../../../rtc_base:rtc_base_approved", ] diff --git a/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.cc b/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.cc index 13a1c2e029..fd11f006eb 100644 --- a/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.cc +++ b/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.cc @@ -13,9 +13,12 @@ #include #include +#include "webrtc/common_types.h" #include "webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h" #include "webrtc/rtc_base/ptr_util.h" #include "webrtc/rtc_base/safe_conversions.h" +#include "webrtc/rtc_base/safe_minmax.h" +#include "webrtc/rtc_base/string_to_number.h" namespace webrtc { namespace { @@ -37,7 +40,22 @@ int GetIlbcBitrate(int ptime) { rtc::Optional AudioEncoderIlbc::SdpToConfig( const SdpAudioFormat& format) { - return AudioEncoderIlbcImpl::SdpToConfig(format); + if (STR_CASE_CMP(format.name.c_str(), "ILBC") != 0 || + format.clockrate_hz != 8000 || format.num_channels != 1) { + return rtc::Optional(); + } + + AudioEncoderIlbcConfig config; + auto ptime_iter = format.parameters.find("ptime"); + if (ptime_iter != format.parameters.end()) { + auto ptime = rtc::StringToNumber(ptime_iter->second); + if (ptime && *ptime > 0) { + const int whole_packets = *ptime / 10; + config.frame_size_ms = rtc::SafeClamp(whole_packets * 10, 20, 60); + } + } + return config.IsOk() ? rtc::Optional(config) + : rtc::Optional(); } void AudioEncoderIlbc::AppendSupportedEncoders( diff --git a/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc index 8bff2ec1ee..711eed73de 100644 --- a/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc +++ b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc @@ -16,7 +16,6 @@ #include "webrtc/common_types.h" #include "webrtc/modules/audio_coding/codecs/g711/g711_interface.h" #include "webrtc/rtc_base/checks.h" -#include "webrtc/rtc_base/string_to_number.h" namespace webrtc { @@ -31,35 +30,6 @@ typename T::Config CreateConfig(const CodecInst& codec_inst) { return config; } -template -typename T::Config CreateConfig(int payload_type, - const SdpAudioFormat& format) { - typename T::Config config; - config.frame_size_ms = 20; - auto ptime_iter = format.parameters.find("ptime"); - if (ptime_iter != format.parameters.end()) { - auto ptime = rtc::StringToNumber(ptime_iter->second); - if (ptime && *ptime > 0) { - const int whole_packets = *ptime / 10; - config.frame_size_ms = std::max(10, std::min(whole_packets * 10, 60)); - } - } - config.num_channels = format.num_channels; - config.payload_type = payload_type; - return config; -} - -template -rtc::Optional QueryAudioEncoderImpl( - const SdpAudioFormat& format) { - if (STR_CASE_CMP(format.name.c_str(), T::GetPayloadName()) == 0 && - format.clockrate_hz == 8000 && format.num_channels >= 1 && - CreateConfig(0, format).IsOk()) { - return rtc::Optional({8000, format.num_channels, 64000}); - } - return rtc::Optional(); -} - } // namespace bool AudioEncoderPcm::Config::IsOk() const { @@ -138,15 +108,6 @@ void AudioEncoderPcm::Reset() { AudioEncoderPcmA::AudioEncoderPcmA(const CodecInst& codec_inst) : AudioEncoderPcmA(CreateConfig(codec_inst)) {} -AudioEncoderPcmA::AudioEncoderPcmA(int payload_type, - const SdpAudioFormat& format) - : AudioEncoderPcmA(CreateConfig(payload_type, format)) {} - -rtc::Optional AudioEncoderPcmA::QueryAudioEncoder( - const SdpAudioFormat& format) { - return QueryAudioEncoderImpl(format); -} - size_t AudioEncoderPcmA::EncodeCall(const int16_t* audio, size_t input_len, uint8_t* encoded) { @@ -164,15 +125,6 @@ AudioEncoder::CodecType AudioEncoderPcmA::GetCodecType() const { AudioEncoderPcmU::AudioEncoderPcmU(const CodecInst& codec_inst) : AudioEncoderPcmU(CreateConfig(codec_inst)) {} -AudioEncoderPcmU::AudioEncoderPcmU(int payload_type, - const SdpAudioFormat& format) - : AudioEncoderPcmU(CreateConfig(payload_type, format)) {} - -rtc::Optional AudioEncoderPcmU::QueryAudioEncoder( - const SdpAudioFormat& format) { - return QueryAudioEncoderImpl(format); -} - size_t AudioEncoderPcmU::EncodeCall(const int16_t* audio, size_t input_len, uint8_t* encoded) { diff --git a/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h index be8164aeee..22a15a1cac 100644 --- a/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h +++ b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h @@ -14,7 +14,6 @@ #include #include "webrtc/api/audio_codecs/audio_encoder.h" -#include "webrtc/api/audio_codecs/audio_format.h" #include "webrtc/rtc_base/constructormagic.h" namespace webrtc { @@ -81,11 +80,6 @@ class AudioEncoderPcmA final : public AudioEncoderPcm { explicit AudioEncoderPcmA(const Config& config) : AudioEncoderPcm(config, kSampleRateHz) {} explicit AudioEncoderPcmA(const CodecInst& codec_inst); - AudioEncoderPcmA(int payload_type, const SdpAudioFormat& format); - - static constexpr const char* GetPayloadName() { return "PCMA"; } - static rtc::Optional QueryAudioEncoder( - const SdpAudioFormat& format); protected: size_t EncodeCall(const int16_t* audio, @@ -110,11 +104,6 @@ class AudioEncoderPcmU final : public AudioEncoderPcm { explicit AudioEncoderPcmU(const Config& config) : AudioEncoderPcm(config, kSampleRateHz) {} explicit AudioEncoderPcmU(const CodecInst& codec_inst); - AudioEncoderPcmU(int payload_type, const SdpAudioFormat& format); - - static constexpr const char* GetPayloadName() { return "PCMU"; } - static rtc::Optional QueryAudioEncoder( - const SdpAudioFormat& format); protected: size_t EncodeCall(const int16_t* audio, diff --git a/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc b/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc index f936e815a7..4c3e82d037 100644 --- a/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc +++ b/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc @@ -17,7 +17,6 @@ #include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h" #include "webrtc/rtc_base/checks.h" #include "webrtc/rtc_base/safe_conversions.h" -#include "webrtc/rtc_base/string_to_number.h" namespace webrtc { @@ -34,27 +33,6 @@ AudioEncoderG722Config CreateConfig(const CodecInst& codec_inst) { } // namespace -rtc::Optional AudioEncoderG722Impl::SdpToConfig( - const SdpAudioFormat& format) { - if (STR_CASE_CMP(format.name.c_str(), "g722") != 0 || - format.clockrate_hz != 8000) { - return rtc::Optional(); - } - - AudioEncoderG722Config config; - config.num_channels = rtc::dchecked_cast(format.num_channels); - auto ptime_iter = format.parameters.find("ptime"); - if (ptime_iter != format.parameters.end()) { - auto ptime = rtc::StringToNumber(ptime_iter->second); - if (ptime && *ptime > 0) { - const int whole_packets = *ptime / 10; - config.frame_size_ms = std::max(10, std::min(whole_packets * 10, 60)); - } - } - return config.IsOk() ? rtc::Optional(config) - : rtc::Optional(); -} - AudioEncoderG722Impl::AudioEncoderG722Impl(const AudioEncoderG722Config& config, int payload_type) : num_channels_(config.num_channels), @@ -78,26 +56,8 @@ AudioEncoderG722Impl::AudioEncoderG722Impl(const AudioEncoderG722Config& config, AudioEncoderG722Impl::AudioEncoderG722Impl(const CodecInst& codec_inst) : AudioEncoderG722Impl(CreateConfig(codec_inst), codec_inst.pltype) {} -AudioEncoderG722Impl::AudioEncoderG722Impl(int payload_type, - const SdpAudioFormat& format) - : AudioEncoderG722Impl(*SdpToConfig(format), payload_type) {} - AudioEncoderG722Impl::~AudioEncoderG722Impl() = default; -rtc::Optional AudioEncoderG722Impl::QueryAudioEncoder( - const SdpAudioFormat& format) { - if (STR_CASE_CMP(format.name.c_str(), GetPayloadName()) == 0) { - const auto config_opt = SdpToConfig(format); - if (format.clockrate_hz == 8000 && config_opt) { - RTC_DCHECK(config_opt->IsOk()); - return rtc::Optional( - {rtc::dchecked_cast(kSampleRateHz), - rtc::dchecked_cast(config_opt->num_channels), 64000}); - } - } - return rtc::Optional(); -} - int AudioEncoderG722Impl::SampleRateHz() const { return kSampleRateHz; } diff --git a/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h b/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h index 00d1c0f3e6..b4d9ef01e4 100644 --- a/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h +++ b/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h @@ -14,7 +14,6 @@ #include #include "webrtc/api/audio_codecs/audio_encoder.h" -#include "webrtc/api/audio_codecs/audio_format.h" #include "webrtc/api/audio_codecs/g722/audio_encoder_g722_config.h" #include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h" #include "webrtc/rtc_base/buffer.h" @@ -26,18 +25,10 @@ struct CodecInst; class AudioEncoderG722Impl final : public AudioEncoder { public: - static rtc::Optional SdpToConfig( - const SdpAudioFormat& format); - AudioEncoderG722Impl(const AudioEncoderG722Config& config, int payload_type); explicit AudioEncoderG722Impl(const CodecInst& codec_inst); - AudioEncoderG722Impl(int payload_type, const SdpAudioFormat& format); ~AudioEncoderG722Impl() override; - static constexpr const char* GetPayloadName() { return "G722"; } - static rtc::Optional QueryAudioEncoder( - const SdpAudioFormat& format); - int SampleRateHz() const override; size_t NumChannels() const override; int RtpTimestampRateHz() const override; diff --git a/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc b/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc index 5a0090ab23..2a6dda7f47 100644 --- a/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc +++ b/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc @@ -16,7 +16,6 @@ #include "webrtc/modules/audio_coding/codecs/ilbc/ilbc.h" #include "webrtc/rtc_base/checks.h" #include "webrtc/rtc_base/safe_conversions.h" -#include "webrtc/rtc_base/string_to_number.h" namespace webrtc { @@ -47,26 +46,6 @@ int GetIlbcBitrate(int ptime) { } // namespace -rtc::Optional AudioEncoderIlbcImpl::SdpToConfig( - const SdpAudioFormat& format) { - if (STR_CASE_CMP(format.name.c_str(), "ilbc") != 0 || - format.clockrate_hz != 8000 || format.num_channels != 1) { - return rtc::Optional(); - } - - AudioEncoderIlbcConfig config; - auto ptime_iter = format.parameters.find("ptime"); - if (ptime_iter != format.parameters.end()) { - auto ptime = rtc::StringToNumber(ptime_iter->second); - if (ptime && *ptime > 0) { - const int whole_packets = *ptime / 10; - config.frame_size_ms = std::max(20, std::min(whole_packets * 10, 60)); - } - } - return config.IsOk() ? rtc::Optional(config) - : rtc::Optional(); -} - AudioEncoderIlbcImpl::AudioEncoderIlbcImpl(const AudioEncoderIlbcConfig& config, int payload_type) : frame_size_ms_(config.frame_size_ms), @@ -81,29 +60,10 @@ AudioEncoderIlbcImpl::AudioEncoderIlbcImpl(const AudioEncoderIlbcConfig& config, AudioEncoderIlbcImpl::AudioEncoderIlbcImpl(const CodecInst& codec_inst) : AudioEncoderIlbcImpl(CreateConfig(codec_inst), codec_inst.pltype) {} -AudioEncoderIlbcImpl::AudioEncoderIlbcImpl(int payload_type, - const SdpAudioFormat& format) - : AudioEncoderIlbcImpl(*SdpToConfig(format), payload_type) {} - AudioEncoderIlbcImpl::~AudioEncoderIlbcImpl() { RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderFree(encoder_)); } -rtc::Optional AudioEncoderIlbcImpl::QueryAudioEncoder( - const SdpAudioFormat& format) { - if (STR_CASE_CMP(format.name.c_str(), GetPayloadName()) == 0) { - const auto config_opt = SdpToConfig(format); - if (format.clockrate_hz == 8000 && format.num_channels == 1 && - config_opt) { - RTC_DCHECK(config_opt->IsOk()); - return rtc::Optional( - {rtc::dchecked_cast(kSampleRateHz), 1, - GetIlbcBitrate(config_opt->frame_size_ms)}); - } - } - return rtc::Optional(); -} - int AudioEncoderIlbcImpl::SampleRateHz() const { return kSampleRateHz; } diff --git a/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h b/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h index c83be61fe9..6a80a69d32 100644 --- a/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h +++ b/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h @@ -12,7 +12,6 @@ #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_ #include "webrtc/api/audio_codecs/audio_encoder.h" -#include "webrtc/api/audio_codecs/audio_format.h" #include "webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc_config.h" #include "webrtc/modules/audio_coding/codecs/ilbc/ilbc.h" #include "webrtc/rtc_base/constructormagic.h" @@ -23,18 +22,10 @@ struct CodecInst; class AudioEncoderIlbcImpl final : public AudioEncoder { public: - static rtc::Optional SdpToConfig( - const SdpAudioFormat& format); - AudioEncoderIlbcImpl(const AudioEncoderIlbcConfig& config, int payload_type); explicit AudioEncoderIlbcImpl(const CodecInst& codec_inst); - AudioEncoderIlbcImpl(int payload_type, const SdpAudioFormat& format); ~AudioEncoderIlbcImpl() override; - static constexpr const char* GetPayloadName() { return "ILBC"; } - static rtc::Optional QueryAudioEncoder( - const SdpAudioFormat& format); - int SampleRateHz() const override; size_t NumChannels() const override; size_t Num10MsFramesInNextPacket() const override; diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h index 2cc23db9fe..c12d734f7c 100644 --- a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h +++ b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h @@ -14,7 +14,6 @@ #include #include "webrtc/api/audio_codecs/audio_encoder.h" -#include "webrtc/api/audio_codecs/audio_format.h" #include "webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h" #include "webrtc/rtc_base/constructormagic.h" #include "webrtc/rtc_base/scoped_ref_ptr.h" @@ -56,13 +55,8 @@ class AudioEncoderIsacT final : public AudioEncoder { explicit AudioEncoderIsacT( const CodecInst& codec_inst, const rtc::scoped_refptr& bwinfo); - AudioEncoderIsacT(int payload_type, const SdpAudioFormat& format); ~AudioEncoderIsacT() override; - static constexpr const char* GetPayloadName() { return "ISAC"; } - static rtc::Optional QueryAudioEncoder( - const SdpAudioFormat& format); - int SampleRateHz() const override; size_t NumChannels() const override; size_t Num10MsFramesInNextPacket() const override; diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h index bda379c8d2..854f2eef0c 100644 --- a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h +++ b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h @@ -13,14 +13,8 @@ #include "webrtc/common_types.h" #include "webrtc/rtc_base/checks.h" -#include "webrtc/rtc_base/string_to_number.h" namespace webrtc { -namespace { // NOLINT (not a "regular" header file) -int GetIsacMaxBitrate(int clockrate_hz) { - return (clockrate_hz == 32000) ? 56000 : 32000; -} -} // namespace template typename AudioEncoderIsacT::Config CreateIsacConfig( @@ -38,33 +32,6 @@ typename AudioEncoderIsacT::Config CreateIsacConfig( return config; } -template -typename AudioEncoderIsacT::Config CreateIsacConfig( - int payload_type, - const SdpAudioFormat& format) { - typename AudioEncoderIsacT::Config config; - config.payload_type = payload_type; - config.sample_rate_hz = format.clockrate_hz; - - // We only support different frame sizes at 16000 Hz. - if (config.sample_rate_hz == 16000) { - auto ptime_iter = format.parameters.find("ptime"); - if (ptime_iter != format.parameters.end()) { - auto ptime = rtc::StringToNumber(ptime_iter->second); - if (ptime && *ptime >= 60) { - config.frame_size_ms = 60; - } else { - config.frame_size_ms = 30; - } - } - } - - // Set the default bitrate for ISAC to the maximum bitrate allowed at this - // clockrate. At this point, adaptive mode is not used by WebRTC. - config.bit_rate = GetIsacMaxBitrate(format.clockrate_hz); - return config; -} - template bool AudioEncoderIsacT::Config::IsOk() const { if (max_bit_rate < 32000 && max_bit_rate != -1) @@ -105,25 +72,6 @@ AudioEncoderIsacT::AudioEncoderIsacT( const rtc::scoped_refptr& bwinfo) : AudioEncoderIsacT(CreateIsacConfig(codec_inst, bwinfo)) {} -template -AudioEncoderIsacT::AudioEncoderIsacT(int payload_type, - const SdpAudioFormat& format) - : AudioEncoderIsacT(CreateIsacConfig(payload_type, format)) {} - -template -rtc::Optional AudioEncoderIsacT::QueryAudioEncoder( - const SdpAudioFormat& format) { - if (STR_CASE_CMP(format.name.c_str(), GetPayloadName()) == 0) { - Config config = CreateIsacConfig(0, format); - if (config.IsOk()) { - return rtc::Optional( - {config.sample_rate_hz, 1, config.bit_rate, 10000, - GetIsacMaxBitrate(format.clockrate_hz)}); - } - } - return rtc::Optional(); -} - template AudioEncoderIsacT::~AudioEncoderIsacT() { RTC_CHECK_EQ(0, T::Free(isac_state_)); diff --git a/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc b/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc index 897eed23f4..7b4a919d7c 100644 --- a/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc +++ b/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc @@ -16,7 +16,6 @@ #include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h" #include "webrtc/rtc_base/checks.h" #include "webrtc/rtc_base/safe_conversions.h" -#include "webrtc/rtc_base/string_to_number.h" namespace webrtc { @@ -35,6 +34,7 @@ AudioEncoder::CodecType AudioEncoderPcm16B::GetCodecType() const { } namespace { + AudioEncoderPcm16B::Config CreateConfig(const CodecInst& codec_inst) { AudioEncoderPcm16B::Config config; config.num_channels = codec_inst.channels; @@ -45,23 +45,6 @@ AudioEncoderPcm16B::Config CreateConfig(const CodecInst& codec_inst) { return config; } -AudioEncoderPcm16B::Config CreateConfig(int payload_type, - const SdpAudioFormat& format) { - AudioEncoderPcm16B::Config config; - config.num_channels = format.num_channels; - config.sample_rate_hz = format.clockrate_hz; - config.frame_size_ms = 10; - auto ptime_iter = format.parameters.find("ptime"); - if (ptime_iter != format.parameters.end()) { - auto ptime = rtc::StringToNumber(ptime_iter->second); - if (ptime && *ptime > 0) { - const int whole_packets = *ptime / 10; - config.frame_size_ms = std::max(10, std::min(whole_packets * 10, 60)); - } - } - config.payload_type = payload_type; - return config; -} } // namespace bool AudioEncoderPcm16B::Config::IsOk() const { @@ -74,24 +57,4 @@ bool AudioEncoderPcm16B::Config::IsOk() const { AudioEncoderPcm16B::AudioEncoderPcm16B(const CodecInst& codec_inst) : AudioEncoderPcm16B(CreateConfig(codec_inst)) {} -AudioEncoderPcm16B::AudioEncoderPcm16B(int payload_type, - const SdpAudioFormat& format) - : AudioEncoderPcm16B(CreateConfig(payload_type, format)) {} - -rtc::Optional AudioEncoderPcm16B::QueryAudioEncoder( - const SdpAudioFormat& format) { - if (STR_CASE_CMP(format.name.c_str(), GetPayloadName()) == 0 && - format.num_channels >= 1) { - Config config = CreateConfig(0, format); - if (config.IsOk()) { - constexpr int bits_per_sample = 16; - return rtc::Optional( - {config.sample_rate_hz, config.num_channels, - config.sample_rate_hz * bits_per_sample * - rtc::dchecked_cast(config.num_channels)}); - } - } - return rtc::Optional(); -} - } // namespace webrtc diff --git a/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h b/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h index 79f56c9788..25d548c0c3 100644 --- a/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h +++ b/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h @@ -31,11 +31,6 @@ class AudioEncoderPcm16B final : public AudioEncoderPcm { explicit AudioEncoderPcm16B(const Config& config) : AudioEncoderPcm(config, config.sample_rate_hz) {} explicit AudioEncoderPcm16B(const CodecInst& codec_inst); - AudioEncoderPcm16B(int payload_type, const SdpAudioFormat& format); - - static constexpr const char* GetPayloadName() { return "L16"; } - static rtc::Optional QueryAudioEncoder( - const SdpAudioFormat& format); protected: size_t EncodeCall(const int16_t* audio,