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for #299, refine the codec object name
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parent
d7458c4e72
commit
caf69f193d
29 changed files with 731 additions and 699 deletions
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@ -278,7 +278,7 @@ int SrsRawH264Stream::mux_avc2flv(string video, int8_t frame_type, int8_t avc_pa
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// Frame Type, Type of video frame.
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// CodecID, Codec Identifier.
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// set the rtmp header
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*p++ = (frame_type << 4) | SrsCodecVideoAVC;
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*p++ = (frame_type << 4) | SrsVideoCodecIdAVC;
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// AVCPacketType
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*p++ = avc_packet_type;
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@ -424,7 +424,7 @@ int SrsRawAacStream::adts_demux(SrsBuffer* stream, char** pframe, int* pnb_frame
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// the codec info.
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codec.protection_absent = protection_absent;
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codec.aac_object = srs_codec_aac_ts2rtmp((SrsAacProfile)profile);
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codec.aac_object = srs_aac_ts2rtmp((SrsAacProfile)profile);
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codec.sampling_frequency_index = sampling_frequency_index;
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codec.channel_configuration = channel_configuration;
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codec.frame_length = frame_length;
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@ -433,15 +433,15 @@ int SrsRawAacStream::adts_demux(SrsBuffer* stream, char** pframe, int* pnb_frame
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// TODO: FIXME: maybe need to resample audio.
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codec.sound_format = 10; // AAC
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if (sampling_frequency_index <= 0x0c && sampling_frequency_index > 0x0a) {
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codec.sound_rate = SrsCodecAudioSampleRate5512;
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codec.sound_rate = SrsAudioSampleRate5512;
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} else if (sampling_frequency_index <= 0x0a && sampling_frequency_index > 0x07) {
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codec.sound_rate = SrsCodecAudioSampleRate11025;
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codec.sound_rate = SrsAudioSampleRate11025;
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} else if (sampling_frequency_index <= 0x07 && sampling_frequency_index > 0x04) {
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codec.sound_rate = SrsCodecAudioSampleRate22050;
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codec.sound_rate = SrsAudioSampleRate22050;
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} else if (sampling_frequency_index <= 0x04) {
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codec.sound_rate = SrsCodecAudioSampleRate44100;
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codec.sound_rate = SrsAudioSampleRate44100;
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} else {
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codec.sound_rate = SrsCodecAudioSampleRate44100;
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codec.sound_rate = SrsAudioSampleRate44100;
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srs_warn("adts invalid sample rate for flv, rate=%#x", sampling_frequency_index);
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}
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codec.sound_type = srs_max(0, srs_min(1, channel_configuration - 1));
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@ -475,11 +475,11 @@ int SrsRawAacStream::mux_sequence_header(SrsRawAacStreamCodec* codec, string& sh
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// override the aac samplerate by user specified.
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// @see https://github.com/ossrs/srs/issues/212#issuecomment-64146899
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switch (codec->sound_rate) {
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case SrsCodecAudioSampleRate11025:
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case SrsAudioSampleRate11025:
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samplingFrequencyIndex = 0x0a; break;
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case SrsCodecAudioSampleRate22050:
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case SrsAudioSampleRate22050:
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samplingFrequencyIndex = 0x07; break;
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case SrsCodecAudioSampleRate44100:
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case SrsAudioSampleRate44100:
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samplingFrequencyIndex = 0x04; break;
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default:
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break;
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@ -532,7 +532,7 @@ int SrsRawAacStream::mux_aac2flv(char* frame, int nb_frame, SrsRawAacStreamCodec
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// 1bytes, SoundFormat|SoundRate|SoundSize|SoundType
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// 1bytes, AACPacketType for SoundFormat == 10, 0 is sequence header.
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int size = nb_frame + 1;
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if (sound_format == SrsCodecAudioAAC) {
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if (sound_format == SrsAudioCodecIdAAC) {
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size += 1;
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}
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char* data = new char[size];
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@ -545,7 +545,7 @@ int SrsRawAacStream::mux_aac2flv(char* frame, int nb_frame, SrsRawAacStreamCodec
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*p++ = audio_header;
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if (sound_format == SrsCodecAudioAAC) {
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if (sound_format == SrsAudioCodecIdAAC) {
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*p++ = aac_packet_type;
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}
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