The　existing　loudness　compensation　algorithms　in　digital　hearing　aids　destroy　the　formant　structure　of　the　speech　signal　easily　and　do　not　consider　the　residue　noise　when　implementing　loudness　compensation　after　speech　enhancement.　As　a　result,　the　output　speech　signal　fails　to　meet　the　requirements　of　hearing-impaired(HI)　people.　To　solve　these　problems,　a　novel　multi-channel　adaptive　loudness　compensation　algorithm　which　can　vary　according　to　signal-to-noise　ratio　(SNR)　is　proposed.　In　this　algorithm,　signals　are　first　divided　into　multiple　frequency　bands　by　the　Gammatone　filter　banks　that　protect　the　formant　structure.　Then,　binary　masked　speech　enhancement　based　on　human　auditory　characteristics　is　implemented　in　each　frequency　band,　removing　noise　as　much　as　possible　while　maintaining　the　authenticity　of　speech.　Essentially,　we　propose　an　adaptive　loudness　compensation　coefficient　which　can　vary　referring　to　the　SNR,　and　adaptively　adjust　the　weight　of　both　the　linear　compensation　and　the　wide　dynamic　compression　in　different　frequency　bands.　The　experimental　results　have　shown　that　compared　with　the　contrast　algorithm,　the　proposed　algorithm　not　only　effectively　protects　the　formant　of　speech　in　the　noise　environment,　but　also　suppresses　the　influence　of　noise　on　the　loudness　performances,　along　with　improvements　in　intelligibility,　comfort　level　and　the　clarity　of　the　speech.