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Fix #2613, Try libopus and opus for ARM (#2614)

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* Fix #2613, Try libopus and opus for ARM

* Refine code
This commit is contained in:
Winlin 2021-09-15 21:44:13 +08:00 committed by GitHub
parent 501b3833a1
commit 560e7c3c2f
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GPG key ID: 4AEE18F83AFDEB23
1 changed files with 30 additions and 15 deletions
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45
trunk/src/app/srs_app_rtc_codec.cpp
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@ -10,16 +10,32 @@
#include <srs_kernel_error.hpp>
#include <srs_kernel_log.hpp>
static const char* id2codec_name(SrsAudioCodecId id)
static const AVCodec* srs_find_decoder_by_id(SrsAudioCodecId id)
{
switch (id) {
case SrsAudioCodecIdAAC:
return "aac";
case SrsAudioCodecIdOpus:
return "libopus";
default:
return "";
if (id == SrsAudioCodecIdAAC) {
return avcodec_find_decoder_by_name("aac");
} else if (id == SrsAudioCodecIdOpus) {
const AVCodec* codec = avcodec_find_decoder_by_name("libopus");
if (!codec) {
codec = avcodec_find_decoder_by_name("opus");
}
return codec;
}
return NULL;
}
static const AVCodec* srs_find_encoder_by_id(SrsAudioCodecId id)
{
if (id == SrsAudioCodecIdAAC) {
return avcodec_find_encoder_by_name("aac");
} else if (id == SrsAudioCodecIdOpus) {
const AVCodec* codec = avcodec_find_encoder_by_name("libopus");
if (!codec) {
codec = avcodec_find_encoder_by_name("opus");
}
return codec;
}
return NULL;
}
class SrsFFmpegLogHelper {
@ -175,10 +191,9 @@ void SrsAudioTranscoder::aac_codec_header(uint8_t **data, int *len)
srs_error_t SrsAudioTranscoder::init_dec(SrsAudioCodecId src_codec)
{
const char* codec_name = id2codec_name(src_codec);
const AVCodec *codec = avcodec_find_decoder_by_name(codec_name);
const AVCodec *codec = srs_find_decoder_by_id(src_codec);
if (!codec) {
return srs_error_new(ERROR_RTC_RTP_MUXER, "Codec not found by name(%d,%s)", src_codec, codec_name);
return srs_error_new(ERROR_RTC_RTP_MUXER, "Codec not found by %d", src_codec);
}
dec_ = avcodec_alloc_context3(codec);
@ -208,15 +223,14 @@ srs_error_t SrsAudioTranscoder::init_dec(SrsAudioCodecId src_codec)
srs_error_t SrsAudioTranscoder::init_enc(SrsAudioCodecId dst_codec, int dst_channels, int dst_samplerate, int dst_bit_rate)
{
const char* codec_name = id2codec_name(dst_codec);
const AVCodec *codec = avcodec_find_encoder_by_name(codec_name);
const AVCodec *codec = srs_find_encoder_by_id(dst_codec);
if (!codec) {
return srs_error_new(ERROR_RTC_RTP_MUXER, "Codec not found by name(%d,%s)", dst_codec, codec_name);
return srs_error_new(ERROR_RTC_RTP_MUXER, "Codec not found by %d", dst_codec);
}
enc_ = avcodec_alloc_context3(codec);
if (!enc_) {
return srs_error_new(ERROR_RTC_RTP_MUXER, "Could not allocate audio codec context(%d,%s)", dst_codec, codec_name);
return srs_error_new(ERROR_RTC_RTP_MUXER, "Could not allocate audio codec context %d", dst_codec);
}
enc_->sample_rate = dst_samplerate;
@ -228,6 +242,7 @@ srs_error_t SrsAudioTranscoder::init_enc(SrsAudioCodecId dst_codec, int dst_chan
if (dst_codec == SrsAudioCodecIdOpus) {
//TODO: for more level setting
enc_->compression_level = 1;
enc_->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
} else if (dst_codec == SrsAudioCodecIdAAC) {
enc_->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
}